[OpenSIPS-Users] SEMS 1.5

[spady]

Hi All, i am trying to get SEMS last rel. but no links seams to work.
Anyone has idea?
Any time I try to search it on iptel web site, NOT FOUND page always
display.

Is this project still maintained?

Regards




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[OpenSIPS-Users] status on t_on_branch

[spady]

Hi all,
I am using opensips 1.11 and using an old code, coming from 1.8, i noticed
that (status=200) does not work.
Is that normal?
Is changed way to use it?

Regards.



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[OpenSIPS-Users] check_source_address not working in 1.9

[spady]

Hi all, i am facing issue on using check_source_address 

Opensips v. : OpenSIPS (1.9.0-notls (i386/linux)) 

Database entries: 

mysql select * from address; 
++-+---+--+--+---+-+--+ 
| id | grp | ip| mask | port | proto | pattern | context_info | 
++-+---+--+--+---+-+--+ 
|  6 |   0 | 10.9.6.3  |   24 |0 | ANY   | |  | 
|  5 |   0 | 172.16.55.201 |   24 |0 | ANY   | |  | 
++-+---+--+--+---+-+--+ 


Subnet Dump: 

root@opensips:./opensipsctl fifo subnet_dump 
   0 0, 10.9.6.0, 255.255.255.0, 0 
   1 0, 172.16.55.0, 255.255.255.0, 0   

Address Relod log; 

Apr 10 10:50:44 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2812]:
DBG:permissions:reload_address_table: number of rows in address table: 2 
Apr 10 10:50:44 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2812]:
DBG:permissions:reload_address_table: Tuple 10.9.6.3, 0, 24, 0 inserted
into subnet table 
Apr 10 10:50:44 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2812]:
DBG:permissions:reload_address_table: Tuple 172.16.55.201, 0, 24, 0
inserted into subnet table 
Apr 10 10:50:44 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2812]:
DBG:permissions:reload_address_table: address table reloaded successfully.



Call logs: 


Apr 10 10:52:00 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2819]:
DBG:permissions:check_src_addr_3: Looking for : 0, 172.16.55.201, 53109, 2
in tables 
Apr 10 10:52:00 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2819]:
DBG:permissions:hash_match: no match in the hash table 
Apr 10 10:52:00 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2819]:
DBG:permissions:match_subnet_table: match found in the subnet table 



Snippet of code: 

if(!check_source_address(0)){ 
if (!proxy_authorize(, subscriber)) { 
proxy_challenge(, 0); 
exit; 


Any idea? 

Regards 



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Re: [OpenSIPS-Users] check_source_address not working in 1.9

[spady]

Hi Bogdan, here snippet of used code:

*if(!check_source_address(0)){
xlog(LOG: Controllo dell'IP sorgente!\n );
if (!proxy_authorize(, subscriber)) {
proxy_challenge(, 0);
exit;
}
}   
if (!db_check_from()) {
sl_send_reply(403,Forbidden auth ID);
exit;
}

consume_credentials();
*

Using this i get *403 Forbidden auth ID*


Using this:

*if(check_source_address(0))*{
xlog(LOG: Controllo dell'IP sorgente!\n );
if (!proxy_authorize(, subscriber)) {
proxy_challenge(, 0);
exit;
}
}   
if (!db_check_from()) {
sl_send_reply(403,Forbidden auth ID);
exit;
}

consume_credentials();

i get:

407 Unauthorized!!






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Re: [OpenSIPS-Users] check_source_address not working with upgrade

[spady]

Hi all, i would arise this post because it's happening same thing to me.

Opensips v. : OpenSIPS (1.9.0-notls (i386/linux))

Database entries:

mysql select * from address;
++-+---+--+--+---+-+--+
| id | grp | ip| mask | port | proto | pattern | context_info |
++-+---+--+--+---+-+--+
|  6 |   0 | 10.9.6.3  |   24 |0 | ANY   | |  |
|  5 |   0 | 172.16.55.201 |   24 |0 | ANY   | |  |
++-+---+--+--+---+-+--+


Subnet Dump:

root@opensips:./opensipsctl fifo subnet_dump
   0 0, 10.9.6.0, 255.255.255.0, 0
   1 0, 172.16.55.0, 255.255.255.0, 0   

Address Relod log;

*Apr 10 10:50:44 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2812]:
DBG:permissions:reload_address_table: number of rows in address table: 2
Apr 10 10:50:44 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2812]:
DBG:permissions:reload_address_table: Tuple 10.9.6.3, 0, 24, 0 inserted
into subnet table
Apr 10 10:50:44 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2812]:
DBG:permissions:reload_address_table: Tuple 172.16.55.201, 0, 24, 0
inserted into subnet table
Apr 10 10:50:44 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2812]:
DBG:permissions:reload_address_table: address table reloaded successfully.*



Call logs:


*Apr 10 10:52:00 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2819]:
DBG:permissions:check_src_addr_3: Looking for : 0, 172.16.55.201, 53109, 2
in tables
Apr 10 10:52:00 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2819]:
DBG:permissions:hash_match: no match in the hash table
Apr 10 10:52:00 opensips
/usr/local/opensips_proxy_1.9.0/sbin/opensips[2819]:
DBG:permissions:match_subnet_table: match found in the subnet table *
   


Snippet of code:

*if(!check_source_address(0)){
if (!proxy_authorize(, subscriber)) {
proxy_challenge(, 0);
exit;
*

Any idea?

Regards





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[OpenSIPS-Users] Modify FROM header with REG EXP

[spady]

Hi all, I am trying to do a weird thing regarding modify FROM header of
INVITE. Let me explain:

This is a snippet of INVITE that i am trying to modify:

*FROM: Name
Surnamesip:0522375542@domain.local;user=phone;epid=6BEB47B1B9;tag=898892e6e7.*

What I want to achieve is:

FROM: Name
Surnamesip:lt;b532542*@domain.local;user=phone;epid=6BEB47B1B9;tag=898892e6e7.

So as you can see i need to modify ONLY a portion of FROM's user name. I
need to strip 05223755, leave 42 and add 5325 prefix.

Is there a way to do that?

I am trying with subst of textops module but i suppose it only works for
R-URI.

uac_replace_from does not give the possibility to use regular expression.

Any idea?

Thanks.




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Re: [OpenSIPS-Users] Help with t_relay

[spady]

Hi Bogdan,
you said:

$ru = sip:user@domain;  # set initial URI 

But the initial URI, in my case, could be different, because is an incoming
call from PSTN towards several DDI users. I can't define ($ru) as i unique
URI. Can I set is as a variable? Am I wrong?

Thanks 



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Re: [OpenSIPS-Users] Help with t_relay

[spady]

Hi all, solved with following solution:

if (src_ip == IP_MEDIANT || (method==INVITE)) { 
rewritehostport(FQDN_IPPBX1:PORT_IPPBX1);
route(10);
exit;
} 


  ..
  ..

route[10] {
append_branch();
rewritehostport(IP_IPPBX2:PORT_IPPBX2);
$du = sip:IP_IPPBX2:PORT_IPPBX2;
# display branches
$(branch(duri)[0]) = sip:IP_IPPBX1:PORT_IPPBX1;transport=tcp; 
  
t_relay();
  exit;
} 

Hope could help someone else.
Regards




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[OpenSIPS-Users] Help with t_relay

[spady]

Hi list, i need some hints by you experts to get working my idea.

What I am trying to do is parallel forking toward 2 different IP PBXs whose
have different carateristics. Let me explain:

IP PBX #1: accept UDP connections on port 5060
IP PBX #2: accept TCP connections on port 5068

What i tried to do is as follow:

if (src_ip == IP_MEDIANT || (method==INVITE)) { 
rewritehostport(FQDN_IP_PBX_2:PORT_IP_PBX_2);
route(10);
exit;
} 

.

route[10] {
append_branch();

  t_relay(tcp:IP_PBX_2:PORT_IP_PBX_2);
  exit;
}

So now I am stuck because i have some dubts that i can't answer:

1- How Can I forward the original request and the new BRANCH created to
different t_relays ( which they have to have different features (UDP, TCp
etc.. ) ) ?

I tried with 

$(branch(uri)[0]) = sip:???@IP_PBX_1:PORT_IP_PBX_1;

But seems not working. In this last snippet of code, Can I use pseudo
variables like $rU??
$(branch(uri)[0]) = sip:$rU@IP_PBX_1:PORT_IP_PBX_1; ( ? )

Hope someone can point me in a right way.

Regards

I did not found into documentation something about protocol for Branches
sections. In my case i also need to change it to UDP.





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Re: [OpenSIPS-Users] Question about Parallel Forking

[spady]

Any Idea??



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[OpenSIPS-Users] Question about Parallel Forking

[spady]

Hi all, i need to do parallel forking towards 2 different systems. When one
of them had picked up the call, opensips has to be able to knows that until
call is terminated. What i need is, if a second call comes again, opensips
has to reply to the caller the busy tone. 
Is this possible?

SCENARIO:


 
E1   SIP   Fork   |
PBX1
PSTN-Mediant 1000---OpenSIPS|
  |
PBX2

Regards



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Re: [OpenSIPS-Users] RTPProxy Problem

[spady]

Hi, take a look at my post. Probably you will find a solution for your issue.

http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-td7581935.html



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Re: [OpenSIPS-Users] RTPProxy Problem

[spady]

Please, post entire opensips log ( set debug to 6 ).

Why are you using same port for sock and notify_sock ??? Have you tried with
different ports?

Bye



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Re: [OpenSIPS-Users] Help: Understanding ACK loop

[spady]

SOLVED!!!

Thanks Bogdan for your hint ;-) That was the problem. PBX was sending a
Contact header NOT compliant to RFC!!!

Just to help someone else, I added this snippet of code:

*if ((status==200)  (src_ip == IP_PBX)) {
 replace(Contact: sip:x.x.x.x.,Contact: sip:x.x.x.x.);
}
} *

Thanks again!!!



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Re: [OpenSIPS-Users] textops and Variable

[spady]

Hi, i really suppose you should use REGEX.
As per manual:

1.3.8.  replace_body_all(re, txt)

Replaces all occurrence of re in the body of the message with txt. Matching
is done on a per-line basis.

Meaning of the parameters is as follows:

*re - Regular expression.*

txt - String.

Regards





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Re: [OpenSIPS-Users] Help: Understanding ACK loop

[spady]

Hi Bogdan, thanks for your time.

Log, in debug 6, is attached as file.

Regards Opnesips_log.txt
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7582829/Opnesips_log.txt
  



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Re: [OpenSIPS-Users] Help to Understand Loop

[spady]

Hi, can someone help me understand this issue? 

Thanks



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[OpenSIPS-Users] Help: Understanding ACK loop

[spady]

Hi, i had to rewrite post because the preview one was too big. Sorry for this
mistake!! Here is the new, short, one:

Hi, thanks for reply. 

Here the SIP dump and after it, configuration. 

IP 172.16.55.100 = Opensips Gw 
IP 10.9.6.3 = PBX 
IP 10.9.6.40 = Gateway GSM 

The problem is when i receive a ACK from Ip 10.9.6.40, Opensips never send
ACK to 10.9.6.3 but instead it sends to itself in a loop way!!!


 2012/10/31 15:01:58.415722 10.9.6.40:5060 - 172.16.55.100:5060 
INVITE sip:2542@172.16.55.100:5060 SIP/2.0. 
From: sip:+@10.9.6.40;tag=q-7313-723a. 
To: sip:2542@172.16.55.100. 
Contact: sip:+@10.9.6.40. 
Call-ID: 1351688617136291@10.9.6.40. 
CSeq: 19584 INVITE. 
Max-Forwards: 70. 
Content-Length: 228. 
Allow: INVITE, BYE, ACK, CANCEL, REGISTER, OPTIONS, REFER, NOTIFY, INFO. 
Record-Route: sip:10.9.6.40. 
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKiekjcfzC14701805113857. 
Content-Type: application/sdp. 
User-Agent: QuesCom SIP Gateway 5.21.028. 
. 
v=0. 
o=QuesCom 1117304981 1117304981 IN IP4 10.9.6.40. 
s=NonSIP. 
c=IN IP4 10.9.6.40. 
t=0 0. 
m=audio 11278 RTP/AVP 18 8 0. 
a=rtpmap:18 g729/8000/1. 
a=fmtp:18 annexb=no. 
a=rtpmap:8 pcma/8000/1. 
a=rtpmap:0 pcmu/8000/1. 
a=sendrecv. 


U 2012/10/31 15:01:58.422423 172.16.55.100:5060 - 10.9.6.40:5060 
SIP/2.0 100 Giving a try. 
From: sip:+@10.9.6.40;tag=q-7313-723a. 
To: sip:2542@172.16.55.100. 
Call-ID: 1351688617136291@10.9.6.40. 
CSeq: 19584 INVITE. 
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKiekjcfzC14701805113857. 
Server: OpenSIPS. 
Content-Length: 0. 
. 


U 2012/10/31 15:01:58.423041 172.16.55.100:5060 - 10.9.6.3:5060 
INVITE sip:2542@10.9.6.3:5060 SIP/2.0. 
Record-Route: sip:172.16.55.100;lr;did=021.929cd6c2. 
From: sip:+@10.9.6.40;tag=q-7313-723a. 
To: sip:2542@172.16.55.100. 
Contact: sip:+@10.9.6.40. 
Call-ID: 1351688617136291@10.9.6.40. 
CSeq: 19584 INVITE. 
Max-Forwards: 69. 
Content-Length: 228. 
Allow: INVITE, BYE, ACK, CANCEL, REGISTER, OPTIONS, REFER, NOTIFY, INFO. 
Record-Route: sip:10.9.6.40. 
Via: SIP/2.0/UDP 172.16.55.100;branch=z9hG4bK1125.b8ca83d.0. 
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKiekjcfzC14701805113857. 
Content-Type: application/sdp. 
User-Agent: QuesCom SIP Gateway 5.21.028. 
. 
v=0. 
o=QuesCom 1117304981 1117304981 IN IP4 10.9.6.40. 
s=NonSIP. 
c=IN IP4 10.9.6.40. 
t=0 0. 
m=audio 11278 RTP/AVP 18 8 0. 
a=rtpmap:18 g729/8000/1. 
a=fmtp:18 annexb=no. 
a=rtpmap:8 pcma/8000/1. 
a=rtpmap:0 pcmu/8000/1. 
a=sendrecv. 


U 2012/10/31 15:01:58.425799 10.9.6.3:5060 - 172.16.55.100:5060 
SIP/2.0 100 Trying. 
To: sip:2542@172.16.55.100. 
From: sip:+@10.9.6.40;tag=q-7313-723a. 
Call-ID: 1351688617136291@10.9.6.40. 
CSeq: 19584 INVITE. 
Via: SIP/2.0/UDP 172.16.55.100;branch=z9hG4bK1125.b8ca83d.0. 
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKiekjcfzC14701805113857. 
Content-Length: 0. 
. 


U 2012/10/31 15:01:58.432723 10.9.6.3:5060 - 172.16.55.100:5060 
SIP/2.0 180 Ringing. 
Record-Route: sip:172.16.55.100;lr;did=021.929cd6c2. 
Record-Route: sip:10.9.6.40. 
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE. 
Contact: sip:10.9.6.3. 
User-Agent: OmniPCX Enterprise R10.0 j1.410.45.a. 
To: sip:2542@172.16.55.100;tag=be6dedc56089bca6b558af187479406f. 
From: sip:+@10.9.6.40;tag=q-7313-723a. 
Call-ID: 1351688617136291@10.9.6.40. 
CSeq: 19584 INVITE. 
Via: SIP/2.0/UDP 172.16.55.100;branch=z9hG4bK1125.b8ca83d.0. 
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKiekjcfzC14701805113857. 
Content-Length: 0. 
. 


U 2012/10/31 15:01:58.434309 172.16.55.100:5060 - 10.9.6.40:5060 
SIP/2.0 180 Ringing. 
Record-Route: sip:172.16.55.100;lr;did=021.929cd6c2. 
Record-Route: sip:10.9.6.40. 
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE. 
Contact: sip:10.9.6.3. 
User-Agent: OmniPCX Enterprise R10.0 j1.410.45.a. 
To: sip:2542@172.16.55.100;tag=be6dedc56089bca6b558af187479406f. 
From: sip:+@10.9.6.40;tag=q-7313-723a. 
Call-ID: 1351688617136291@10.9.6.40. 
CSeq: 19584 INVITE. 
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKiekjcfzC14701805113857. 
Content-Length: 0. 
. 


























U 2012/10/31 15:02:02.908748 10.9.6.3:5060 - 172.16.55.100:5060 
SIP/2.0 200 OK. 
Record-Route: sip:172.16.55.100;lr;did=021.929cd6c2. 
Record-Route: sip:10.9.6.40. 
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE. 
Contact: sip:10.9.6.3. 
Supported: replaces,timer,path,100rel. 
User-Agent: OmniPCX Enterprise R10.0 j1.410.45.a. 
Session-Expires: 1800;refresher=uas. 
P-Asserted-Identity: User A sip:5222542@10.9.6.3;user=phone. 
Content-Type: application/sdp. 
To: sip:2542@172.16.55.100;tag=be6dedc56089bca6b558af187479406f. 
From: sip:+@10.9.6.40;tag=q-7313-723a. 
Call-ID: 1351688617136291@10.9.6.40. 
CSeq: 19584 INVITE. 
Via: SIP/2.0/UDP 172.16.55.100;branch=z9hG4bK1125.b8ca83d.0. 
Via: SIP/2.0/UDP 

Re: [OpenSIPS-Users] Help: Understanding ACK loop

[spady]

Hi Qasim, i tried but same issue. 
SIP trace is changed a bit,like below, but issue not. i also get this errors
into opensips logs but i cannot uderstand them.

U 2012/11/06 20:33:25.046107 10.9.6.3:5060 - 172.16.55.100:5060
SIP/2.0 200 OK.
Record-Route: sip:172.16.55.100;lr;did=2b.a230b8e6.
Record-Route: sip:10.9.6.40.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE.
Contact: sip:10.9.6.3.
Supported: replaces,timer,path,100rel.
User-Agent: OmniPCX Enterprise R10.0 j1.410.45.a.
Session-Expires: 1800;refresher=uas.
P-Asserted-Identity: Mussini Andrea sip:5222541@10.9.6.3;user=phone.
Content-Type: application/sdp.
To: sip:2541@172.16.55.100;tag=4f663091f6d6b5553461b0b4c9c2df35.
From: sip:+393480806946@10.9.6.40;tag=q-7313-a941.
Call-ID: 1352226984035420@10.9.6.40.
CSeq: 25311 INVITE.
Via: SIP/2.0/UDP 172.16.55.100;branch=z9hG4bK7b8e.1a54d8a2.0.
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKcdkyjpbC17142343480756.
Content-Length: 194.
.
v=0.
o=OXE 1352230298 1352230298 IN IP4 10.9.6.3.
s=abs.
c=IN IP4 10.9.6.111.
t=0 0.
m=audio 32514 RTP/AVP 18.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=ptime:30.
a=maxptime:40.


U 2012/11/06 20:33:25.046911 172.16.55.100:5060 - 10.9.6.40:5060
SIP/2.0 200 OK.
Record-Route: sip:172.16.55.100;lr;did=2b.a230b8e6.
Record-Route: sip:10.9.6.40.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE.
Contact: sip:10.9.6.3.
Supported: replaces,timer,path,100rel.
User-Agent: OmniPCX Enterprise R10.0 j1.410.45.a.
Session-Expires: 1800;refresher=uas.
P-Asserted-Identity: Mussini Andrea sip:5222541@10.9.6.3;user=phone.
Content-Type: application/sdp.
To: sip:2541@172.16.55.100;tag=4f663091f6d6b5553461b0b4c9c2df35.
From: sip:+393480806946@10.9.6.40;tag=q-7313-a941.
Call-ID: 1352226984035420@10.9.6.40.
CSeq: 25311 INVITE.
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKcdkyjpbC17142343480756.
Content-Length: 194.
.
v=0.
o=OXE 1352230298 1352230298 IN IP4 10.9.6.3.
s=abs.
c=IN IP4 10.9.6.111.
t=0 0.
m=audio 32514 RTP/AVP 18.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=ptime:30.
a=maxptime:40.


U 2012/11/06 20:33:25.052071 10.9.6.40:5060 - 172.16.55.100:5060
ACK sip:172.16.55.100;lr;did=2b.a230b8e6 SIP/2.0.
From: sip:+393480806946@10.9.6.40;tag=q-7313-a941.
To: sip:2541@172.16.55.100;tag=4f663091f6d6b5553461b0b4c9c2df35.
Call-ID: 1352226984035420@10.9.6.40.
CSeq: 25311 ACK.
Max-Forwards: 70.
Content-Length: 0.
Contact: sip:+393480806946@10.9.6.40.
Route: sip:10.9.6.3.
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKypebcqkC17142343496466.
.


U 2012/11/06 20:33:25.054039 172.16.55.100:5060 - 172.16.55.100:5060
ACK sip:172.16.55.100;lr;did=2b.a230b8e6 SIP/2.0.
From: sip:+393480806946@10.9.6.40;tag=q-7313-a941.
To: sip:2541@172.16.55.100;tag=4f663091f6d6b5553461b0b4c9c2df35.
Call-ID: 1352226984035420@10.9.6.40.
CSeq: 25311 ACK.
Max-Forwards: 69.
Content-Length: 0.
Contact: sip:+393480806946@10.9.6.40.
Route: sip:10.9.6.3.
Via: SIP/2.0/UDP 172.16.55.100;branch=z9hG4bK7b8e.1a54d8a2.2.
Via: SIP/2.0/UDP 10.9.6.40:5060;branch=z9hG4bKypebcqkC17142343496466.
. 

After them, 10.9.6.3 try to send again 200 OK until it sends BYE ( because
it has not recived ACK ).

Here logs:

Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:parse_msg: SIP
Request:
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:parse_msg: 
method:  ACK
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:parse_msg: 
uri: sip:172.16.55.100;lr;did=99d.779fa501
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:parse_msg: 
version: SIP/2.0
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:parse_headers:
flags=2
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]:
DBG:core:parse_to_param: tag=b67d1f231c591c0d3a3ec185f23148eb
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:parse_to: end
of header reached, state=29
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:parse_to:
display={}, ruri={sip:2541@172.16.55.100}
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:get_hdr_field:
To [63]; uri=[sip:2541@172.16.55.100]
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:get_hdr_field:
to body [sip:2541@172.16.55.100]
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:get_hdr_field:
cseq CSeq: 25317 ACK
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: DBG:core:get_hdr_field:
content_length=0
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]:
DBG:core:parse_via_param: found param type 232, branch =
z9hG4bKflkekrjC17172343804901; state=16
Nov  6 20:38:33 opensips
/usr/local/opensips_proxy_1.8.2/sbin/opensips[7865]: 

Re: [OpenSIPS-Users] Help to Understand Loop

[spady]

Him nobody has an idea? I searched for similar issue in the forum and I found
something similar in this treat:

http://opensips-open-sip-server.1449251.n2.nabble.com/loose-route-loop-on-ACK-requests-td2462835.html

I checked but in the domain table there is only opensips ip.

Thanks



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Re: [OpenSIPS-Users] Help to Understand Loop

[spady]

I found in the last 200 OK, before ACK, that contact is like this:

*Contact: sip:10.9.6.3*

but, comparing whit another call, that should be something like:

*Contact: sip:2542@10.9.6.3*

Should be this the issue? For some reason Opensips does not like that format
and inserts into ACK itself instead of the content of contact header.

Should be?

Thanks



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Re: [OpenSIPS-Users] Help to Understand Loop

[spady]

Hi, I found this ERRORS in the log, but i cannot understand why. What Am i
missing?

ov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_msg:
SIP Reply  (status):
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_msg: 
version: SIP/2.0
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_msg: 
status:  200
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_msg: 
reason:  OK
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_headers: flags=2
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_to_param: tag=cf23470740b0c4eda779658d46a1c002
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_to: end
of header reached, state=29
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_to:
display={}, ruri={sip:2541@172.16.55.100}
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:get_hdr_field: To [63]; uri=[sip:2541@172.16.55.100]
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:get_hdr_field: to body [sip:2541@172.16.55.100]
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:get_hdr_field: cseq CSeq: 21399 INVITE
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_via_param: found param type 232, branch =
z9hG4bK539e.4cfb6ce4.0; state=16
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_via:
end of header reached, state=5
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_headers: via found, flags=2
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_headers: this is the first via
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:receive_msg:
After parse_msg...
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:forward_reply: found module tm, passing reply to it
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:tm:t_check:
start=0x
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_headers: flags=22
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_headers: flags=8
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:tm:t_reply_matching: hash 59701 label 1321648068 branch 0
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:tm:t_reply_matching: REF_UNSAFE:[0xaeeb3ee8] after is 1
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:tm:t_reply_matching: reply matched (T=0xaeeb3ee8)!
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_headers: flags=8
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:tm:run_trans_callbacks: trans=0xaeeb3ee8, callback type 2, id 1 entered
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_headers: flags=8
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:tm:t_check:
end=0xaeeb3ee8
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:tm:reply_received:
org. status uas=180, uac[0]=180 local=0 is_invite=1)
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:tm:t_should_relay_response: T_code=180, new_code=200
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:tm:relay_reply:
branch=0, save=0, relay=0
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:tm:run_trans_callbacks: trans=0xaeeb3ee8, callback type 8, id 0 entered
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:dialog:push_reply_in_dialog: 0xaeeb2818 totag in rpl is
cf23470740b0c4eda779658d46a1c002 (32)
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:dialog:push_reply_in_dialog: branch with tag
cf23470740b0c4eda779658d46a1c002 already exists
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_headers: flags=
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_via_param: found param type 232, branch =
z9hG4bKwjtmvdfC15411976037827; state=1
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_via:
end of header reached, state=5
Nov  2 14:30:21 opensips
/usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]:
DBG:core:parse_headers: via found, 

Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

Hi Nick, you SOLVED my issue!!!

Thank you very much.
So, seems rtpproxy's bug is not resolved yet!!

Hope this long treat can help someone else.

regards



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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

Hi all, i am still stuck on this problem and I don't really know how to solve
it.
Is there somebody that can help me understand what's wrong?




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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

Hi Nick, you mean something like this?

rtpproxy -F -s udp:127.0.0.1:10177 -l 10.9.23.41/151.x.x.201 -u root *-n
unix:/var/run/rtpproxy_timeout.sock* -d DBUG:LOG_LOCAL2

The rtpproxy man page ( http://linux.die.net/man/8/rtpproxy ) says
following:

*-n timeout_socket
This parameter configures the optional timeout notification socket. The
socket should be created by another application, preferably before starting
rtpproxy. For those sessions where the timeout mechanism is enabled,
notifications are sent on this socket if the session times out.
Example: -n unix:/var/run/rtpproxy_timeout.sock

There is no default value, notifications are not sent and not permitted
unless a value is specified explicitly.*

The socket should be created by another application. How can I create it? by
what?

Thanks



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Re: [OpenSIPS-Users] how to set $var(name) in opensips.cfg?

[spady]

Hi Bogdan, I am trying to do the same thing but what if in my script I have
different destination uri?

Ex. if ...
route(2);

 if
   route(3);
...

the pv  $du  could be only one, for instance, $du = $var(new_uri); and NOT
*$du1 = $var(new_uri); $du2 = $var(new_uri);* 

Am I wrong?

Regards




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Re: [OpenSIPS-Users] Perl Script

[spady]

Hi Binan, thanks for your hint. I resolved my problem, without using perl
script.
Thanks to everybody.




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Re: [OpenSIPS-Users] Perl Script

[spady]

No one as idea?
Just in case, is there another way to convert 183 into 180?
Thanks



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Re: [OpenSIPS-Users] Perl Script

[spady]

Thanks Muhammad,
i will wait for your reply.
Btw, I am using OpenSIPS v. 1.8.0



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Re: [OpenSIPS-Users] Perl Script

[spady]

Is there a way to test offline the perl script to check what's wrong?
Seems that script can't extract IP from VIA header of 183 message.
Am I wrong?




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[OpenSIPS-Users] Catch 183 Session Progress

[spady]

Hi all, in a trace like this, how can I catch the 183 message? I tried with
this code

if (t_check_status(183)) {
 # no support for early media
   xlog(LOG: Individuo il 183\n );
}

or

if (status==183) {
 # no support for early media
   xlog(LOG: Individuo il 183\n );
}

But never appear on opensips log that i found it. I can't understand where,
in the script, i have to put it
What's wrong?


U 2012/10/15 11:46:48.362832 172.16.52.51:5060 - 172.16.55.100:5060
INVITE sip:3707@172.16.55.100;user=phone SIP/2.0.
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE.
Supported: 100rel,from-change,timer,histinfo.
User-Agent: OXO_GW_820/044.001.
Session-Expires: 43200.
P-Asserted-Identity: Pippo sip:100@172.16.52.51;user=phone.
History-Info: sip:3707@172.16.55.100;user=phone;index=1.
To: sip:3707@172.16.55.100;user=phone.
From: Pippo
sip:100@172.16.52.51;user=phone;tag=2dd6eb349c4b183f08a67d2ad065655e.
Contact: Pippo sip:100@172.16.52.51;transport=UDP;user=phone.
Content-Type: application/sdp.
Call-ID: 3f9d299c479416397996522871b87398@172.16.52.51.
CSeq: 199290652 INVITE.
Via: SIP/2.0/UDP
172.16.52.51;rport;branch=z9hG4bK8deb021f4c19be2ec40d3d277d6a6684.
Max-Forwards: 70.
Content-Length: 215.
.
v=0.
o=default 1350294284 1350294284 IN IP4 172.16.52.51.
s=-.
c=IN IP4 172.16.52.51.
t=0 0.
m=audio 32000 RTP/AVP 8 106 0.
a=sendrecv.
a=rtpmap:106 telephone-event/8000.
a=fmtp:106 0-15.
a=ptime:20.
a=maxptime:90.


U 2012/10/15 11:46:48.366078 172.16.55.100:5060 - 172.16.52.51:5060
SIP/2.0 100 Giving a try.
To: sip:3707@172.16.55.100;user=phone.
From: Pippo
sip:100@172.16.52.51;user=phone;tag=2dd6eb349c4b183f08a67d2ad065655e.
Call-ID: 3f9d299c479416397996522871b87398@172.16.52.51.
CSeq: 199290652 INVITE.
Via: SIP/2.0/UDP
172.16.52.51;received=172.16.52.51;rport=5060;branch=z9hG4bK8deb021f4c19be2ec40d3d277d6a6684.
Server: OpenSIPS-Longwave.
Content-Length: 0.
.


T 2012/10/15 11:46:48.367179 172.16.55.100:44921 - 10.9.101.163:5068 [AP]
INVITE sip:3...@lync.lwtec.eu:5068;user=phone SIP/2.0.
Record-Route: sip:172.16.55.100;transport=tcp;r2=on;lr;did=9c5.414e9ea5.
Record-Route: sip:172.16.55.100;r2=on;lr;did=9c5.414e9ea5.
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE.
Supported: 100rel,from-change,timer,histinfo.
User-Agent: OXO_GW_820/044.001.
Session-Expires: 43200.
P-Asserted-Identity: Pippo sip:100@172.16.52.51;user=phone.
History-Info: sip:3707@172.16.55.100;user=phone;index=1.
To: sip:3707@172.16.55.100;user=phone.
From: Pippo
sip:100@172.16.52.51;user=phone;tag=2dd6eb349c4b183f08a67d2ad065655e.
Contact: Pippo sip:100@172.16.52.51;transport=UDP;user=phone.
Content-Type: application/sdp.
Call-ID: 3f9d299c479416397996522871b87398@172.16.52.51.
CSeq: 199290652 INVITE.
Via: SIP/2.0/TCP 172.16.55.100;branch=z9hG4bKb16.90f5e873.0.
Via: SIP/2.0/UDP
172.16.52.51;received=172.16.52.51;rport=5060;branch=z9hG4bK8deb021f4c19be2ec40d3d277d6a6684.
Max-Forwards: 69.
Content-Length: 215.
.
v=0.
o=default 1350294284 1350294284 IN IP4 172.16.52.51.
s=-.
c=IN IP4 172.16.52.51.
t=0 0.
m=audio 32000 RTP/AVP 8 106 0.
a=sendrecv.
a=rtpmap:106 telephone-event/8000.
a=fmtp:106 0-15.
a=ptime:20.
a=maxptime:90.


T 2012/10/15 11:46:48.368471 10.9.101.163:5068 - 172.16.55.100:44921 [AP]
SIP/2.0 100 Trying.
FROM:
Pipposip:100@172.16.52.51;user=phone;tag=2dd6eb349c4b183f08a67d2ad065655e.
TO: sip:3707@172.16.55.100;user=phone.
CSEQ: 199290652 INVITE.
CALL-ID: 3f9d299c479416397996522871b87398@172.16.52.51.
VIA: SIP/2.0/TCP 172.16.55.100;branch=z9hG4bKb16.90f5e873.0,SIP/2.0/UDP
172.16.52.51;received=172.16.52.51;branch=z9hG4bK8deb021f4c19be2ec40d3d277d6a6684;rport=5060.
CONTENT-LENGTH: 0.
.


T 2012/10/15 11:46:49.073898 10.9.101.163:5068 - 172.16.55.100:44921 [AP]
SIP/2.0 183 Session Progress.
FROM:
Pipposip:100@172.16.52.51;user=phone;tag=2dd6eb349c4b183f08a67d2ad065655e.
TO: sip:3707@172.16.55.100;user=phone;tag=5a974d1727;epid=D430E933C4.
CSEQ: 199290652 INVITE.
CALL-ID: 3f9d299c479416397996522871b87398@172.16.52.51.
VIA: SIP/2.0/TCP 172.16.55.100;branch=z9hG4bKb16.90f5e873.0,SIP/2.0/UDP
172.16.52.51;received=172.16.52.51;branch=z9hG4bK8deb021f4c19be2ec40d3d277d6a6684;rport=5060.
RECORD-ROUTE:
sip:172.16.55.100;transport=tcp;r2=on;lr;did=9c5.414e9ea5,sip:172.16.55.100;r2=on;lr;did=9c5.414e9ea5.
CONTACT: sip:LYNC.lwtec.eu:5068;transport=Tcp;maddr=10.9.101.163.
CONTENT-LENGTH: 255.
CONTENT-TYPE: application/sdp.
ALLOW: CANCEL.
ALLOW: BYE.
ALLOW: UPDATE.
ALLOW: PRACK.
REQUIRE: 100rel.
SERVER: RTCC/4.0.0.0 MediationServer.
Rseq: 1.
.
v=0.
o=- 322 1 IN IP4 10.9.101.163.
s=session.
c=IN IP4 10.9.101.163.
b=CT:1000.
t=0 0.
m=audio 49932 RTP/AVP 8 106.
c=IN IP4 10.9.101.163.
a=rtcp:49933.
a=label:Audio.
a=sendrecv.
a=rtpmap:8 PCMA/8000.
a=rtpmap:106 telephone-event/8000.
a=fmtp:106 0-16.


U 2012/10/15 11:46:49.075356 172.16.55.100:5060 - 172.16.52.51:5060
SIP/2.0 183 Session Progress.
FROM:

[OpenSIPS-Users] Perl Script

[spady]

Hi all, I am trying to use the perl script that is present under tutorial
section of opensips web site but i have and error on opensips logs and i
don't know why, here is the log:

*Oct 15 15:11:14 opensips /usr/local/opensips_proxy/sbin/opensips[28780]:
ERROR:core:XS_OpenSIPS__Message_log: perl error: Can't locate object method
new via package IO::Socket::INET (perhaps you forgot to load
IO::Socket::INET?) at
/usr/local/opensips_proxy/etc/opensips/perlfunctions.pl line 134.#012 *

and here is the snippet of code involved :

*ub sendSipMessage {
my $ip = shift;
my $port = shift;
my $msg = shift;
my $sock = new IO::Socket::INET (
PeerAddr  = $ip, 
PeerPort  = $port,
Proto = 'udp',
LocalPort = '5060',
ReuseAddr = '1'
);
*

Have i to modify the line my $sock = new IO::Socket::INET ?? or i really
missed to load something?
Sorry, for dummy question, but i searched on forum but there is nothing that
give me a clue.

Thanks



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Re: [OpenSIPS-Users] Perl Script

[spady]

Hi Brett, here is the code:

use OpenSIPS qw ( log );
use OpenSIPS::Constants;

###
# Create a hashref out of ab=123;bc=45
##
sub splitKeyValue {
my @parts = split /\;/, shift;
my $avp;
my $key;
my $val;
while (my $part = shift(@parts)) {
($key, $val) = split /=/, $part, 2;
$avp-{$key} = $val;
}
return $avp;
}

###
# Return a hashref of arrays with all headers found in given string,
# grouped by header name (case sensitive!)
##
sub parseHeaderLines {
my $header = shift;
my @lines = split /\r?\n/, $header;
my $headers;
my $key;
my $val;
while ($line = shift @lines) {
($key, $val) = split /:\s*/, $line, 2;
my @values = split /,/, $val;
push @{$headers-{$key}}, @values;
}
return $headers;
}

###
# Should be called for 183 replies, that need to be converted to
# SDP-less 180 Ringing replies
##
sub sendReplyAs180 {
my $vias;
my $via;
my $via_params;
my $top_via;
my $new_header;
my $headers;
my $status_line;
my $port = 5060;
my $message = shift;
my @header_lines = split /\r\n/, $message-getFullHeader();

# Separate Via lines from the rest of the header
foreach (@header_lines) {
if (/^Via:/) {
$via .= $_ . \r\n;
} else {
if (! $status_line) {
$status_line = $_ . \r\n;
} else {
# Skip Content-* lines
$headers .= $_ . \r\n if ! /^Content-/i;
}
}
}

# Add Content-Length: 0
$headers .= Content-Length: 0\r\n\r\n;

# Start new header with different status line
$new_header = SIP/2.0 180 Ringing\r\n;

# Remove topmost Via
$vias = parseHeaderLines($via);
shift @{$vias-{Via}};
foreach $key (keys %$vias) {
# Add remaining Via's to new header
foreach (@{$vias-{$key}}) {
$new_header .= Via: $_\r\n;
}
}

# Re-add other headers
$new_header .= $headers;

# Retrieve destination ip and port, with respect to received and rport
$top_via = $vias-{Via}[0];
($dummy, $top_via) = split /\s+/, $top_via, 2;
($ip, $top_via) = split /;/, $top_via, 2;
my $via_params = splitKeyValue($top_via);
if ($ip =~ /^(.+)\:(.+)$/) {
$ip = $1;
$port = $2;
}
$ip = $via_params-{received} if $via_params-{received} =~
/^[0-9\.]+$/;
$port = $via_params-{rport} if $via_params-{rport} =~ /^\d{4,5}$/;

# Finally send out the packet
log(L_INFO, Sending reply transformed to 180 Ringing to $ip:$port);
sendSipMessage($ip, $port, $new_header);
return 1;
}

###
# Send a given SIP message to given IP and port
##
sub sendSipMessage {
my $ip = shift;
my $port = shift;
my $msg = shift;
my $sock = new IO::Socket::INET (
PeerAddr  = $ip, 
PeerPort  = $port,
Proto = 'udp',
LocalPort = '5060',
ReuseAddr = '1'
);
return unless $sock;
print $sock $msg;
close($sock);
}




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Re: [OpenSIPS-Users] Perl Script

[spady]

Tried but when restart Opensips I get this error:

*Oct 15 15:46:31 opensips /usr/local/opensips_proxy/sbin/opensips[29672]:
ERROR:core:XS_OpenSIPS__Message_log: perl error: Can't locate object method
Use via package IO::Socket (perhaps you forgot to load IO::Socket?) at
/usr/local/opensips_proxy/etc/opensips/perlfunctions.pl line 26.#012 *





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Re: [OpenSIPS-Users] Perl Script

[spady]

Ok Brett!!
That's was the problem. 
Thanks a lot 



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Re: [OpenSIPS-Users] Perl Script

[spady]

Hi Brett and all, after some tests i can use the perl script but it has to be
adjusted to fit my enviroment.
Now i see on opensips log that the $ip is missing, infact i have this error:

*Oct 15 16:48:17 opensips /usr/local/opensips_proxy/sbin/opensips[31821]:
INFO:core:XS_OpenSIPS_log: Sending reply transformed to 180 Ringing to :5060
Oct 15 16:48:17 opensips /usr/local/opensips_proxy/sbin/opensips[31821]:
ERROR:core:parse_uri: uri too short: 183 (3)
Oct 15 16:48:17 opensips /usr/local/opensips_proxy/sbin/opensips[31821]:
ERROR:core:do_action: bad uri 183, dropping packet
Oct 15 16:48:17 opensips /usr/local/opensips_proxy/sbin/opensips[31821]:
CRITICAL:tm:w_t_relay: unsupported route type: 4   
*

As you can see, the IP is not inserted.

Here is the entire perl code used:

*
use OpenSIPS qw ( log );
use OpenSIPS::Constants;
use IO::Socket;

###
# Create a hashref out of ab=123;bc=45
##
sub splitKeyValue {
my @parts = split /\;/, shift;
my $avp;
my $key;
my $val;
while (my $part = shift(@parts)) {
($key, $val) = split /=/, $part, 2;
$avp-{$key} = $val;
}
return $avp;
}

###
# Return a hashref of arrays with all headers found in given string,
# grouped by header name (case sensitive!)
##
sub parseHeaderLines {
my $header = shift;
my @lines = split /\r?\n/, $header;
my $headers;
my $key;
my $val;
while ($line = shift @lines) {
($key, $val) = split /:\s*/, $line, 2;
my @values = split /,/, $val;
push @{$headers-{$key}}, @values;
}
return $headers;
}

###
# Should be called for 183 replies, that need to be converted to
# SDP-less 180 Ringing replies
##
sub sendReplyAs180 {
my $vias;
my $via;
my $via_params;
my $top_via;
my $new_header;
my $headers;
my $status_line;
my $port = 5060;
my $message = shift;
my @header_lines = split /\r\n/, $message-getFullHeader();

# Separate Via lines from the rest of the header
foreach (@header_lines) {
if (/^Via:/) {
$via .= $_ . \r\n;
} else {
if (! $status_line) {
$status_line = $_ . \r\n;
} else {
# Skip Content-* lines
$headers .= $_ . \r\n if ! /^Content-/i;
}
}
}

# Add Content-Length: 0
$headers .= Content-Length: 0\r\n\r\n;

# Start new header with different status line
$new_header = SIP/2.0 180 Ringing\r\n;

# Remove topmost Via
$vias = parseHeaderLines($via);
shift @{$vias-{Via}};
foreach $key (keys %$vias) {
# Add remaining Via's to new header
foreach (@{$vias-{$key}}) {
$new_header .= Via: $_\r\n;
}
}

# Re-add other headers
$new_header .= $headers;

# Retrieve destination ip and port, with respect to received and rport
$top_via = $vias-{Via}[0];
($dummy, $top_via) = split /\s+/, $top_via, 2;
($ip, $top_via) = split /;/, $top_via, 2;
my $via_params = splitKeyValue($top_via);
if ($ip =~ /^(.+)\:(.+)$/) {
$ip = $1;
$port = $2;
}
$ip = $via_params-{received} if $via_params-{received} =~
/^[0-9\.]+$/;
$port = $via_params-{rport} if $via_params-{rport} =~ /^\d{4,5}$/;

# Finally send out the packet
log(L_INFO, Sending reply transformed to 180 Ringing to $ip:$port);
sendSipMessage($ip, $port, $new_header);
return 1;
}

###
# Send a given SIP message to given IP and port
##
sub sendSipMessage {
my $ip = shift;
my $port = shift;
my $msg = shift;
my $sock = new IO::Socket::INET (
PeerAddr  = $ip, 
PeerPort  = $port,
Proto = 'udp',
LocalPort = '5060',
ReuseAddr = '1'
);
return unless $sock;
print $sock $msg;
close($sock);
}
*

Is there a way to output in some logs the builded new SIP MESSAGE? In
opensips log i can only see the error log but not how is builded.

Why $ip results null???
Thanks



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Re: [OpenSIPS-Users] dlg_validate_dialog Error, help

[spady]

Any idea?



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Re: [OpenSIPS-Users] dlg_validate_dialog Error, help

[spady]

Hi Vlad, thanks for reply.
I am trying to follow your hint but I get this error when I start Opensips:

*Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t: has_totag
not found
Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t: found
has_totag(0) in module uri
[/usr/local/opensips_proxy/lib/opensips/modules/]
Oct 11 10:29:45 opensips opensips: CRITICAL:core:yyerror: parse error in
config file
/usr/local/opensips_proxy/etc/opensips/opensips_residential_2012-6-11_14:13:47.CFG,
line 532, column 17-18: Command cannot be used in the block#012
Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t: loose_route
not found
Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t: found
loose_route(0) in module rr
[/usr/local/opensips_proxy/lib/opensips/modules/]
Oct 11 10:29:45 opensips opensips: CRITICAL:core:yyerror: parse error in
config file
/usr/local/opensips_proxy/etc/opensips/opensips_residential_2012-6-11_14:13:47.CFG,
line 533, column 21-22: Command cannot be used in the block#012
Oct 11 10:29:45 opensips opensips: DBG:core:pv_lookup_spec_name: found in
extra list [DLG_status]
Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t:
validate_dialog not found
Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t: found
validate_dialog(0) in module dialog
[/usr/local/opensips_proxy/lib/opensips/modules/]
Oct 11 10:29:45 opensips opensips: CRITICAL:core:yyerror: parse error in
config file
/usr/local/opensips_proxy/etc/opensips/opensips_residential_2012-6-11_14:13:47.CFG,
line 535, column 34-35: Command cannot be used in the block#012
Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t:
fix_route_dialog not found
Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t: found
fix_route_dialog(0) in module dialog
[/usr/local/opensips_proxy/lib/opensips/modules/]
Oct 11 10:29:45 opensips opensips: CRITICAL:core:yyerror: parse error in
config file
/usr/local/opensips_proxy/etc/opensips/opensips_residential_2012-6-11_14:13:47.CFG,
line 536, column 34-35: Command cannot be used in the block#012 
  
*


Here is the code:


branch_route[2] {   
if (is_method(INVITE)  is_audio_on_hold()){
if (has_totag()) {
loose_route();
if ($DLG_status!=NULL)
if (!validate_dialog())
fix_route_dialog();
   }
 set_dlg_flag(7);
  } else {
   if (is_method(INVITE)){
reset_dlg_flag(7);
  }
} 
if (is_dlg_flag_set(7)  (status==200)){
replace_body(a=sendrecv.,a=inactive);
  }   
xlog(new branch at $ru\n);
 }   



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Re: [OpenSIPS-Users] dlg_validate_dialog Error, help

[spady]

Hi Vlad. Ok now it's a bit clear. 
What i am trying to do is to solve the following issue:

PBX---OpensipsLync server

Opensips acts as UDP/TCP proxy. Call can go from PBX to LYNC and viceversa.
The problem comes when from LYNC I put on hold the call. Everytime i get the
following error, just after i press hold button on LYNC:

*Oct 11 13:43:04 opensips /usr/local/opensips_proxy/sbin/opensips[9512]:
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
dlg=[sip:5100@172.16.52.51;transport=UDP;user=phone] ,
req=[sip:5100@172.16.52.51;user=phone]*

So I want to solve it!!

Now I tried to put  fix_route_dialog()  in sequential requestes but
problem is still present. Maybe i did not understand where to put it or how
to use it. Can you help me?

if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {

# validate the sequential request against dialog
if ( $DLG_status!=NULL  !validate_dialog() ) {
xlog(In-Dialog $rm from $si (callid=$ci) is 
not valid according to
dialog\n);
## exit;
}

if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless, do 
RR for
# re-INVITEs alos, as some buggy clients do 
change route set
# during the dialog.
record_route();
*fix_route_dialog();*

}



Thanks




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Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

[spady]

SOLVED!!
It was because i have an error on dialog behaviour. For now I solved in this
way, hope could be interesting for someone else:

*branch_route[2] {  
if (is_method(INVITE)  is_audio_on_hold()){
if ( search_body(a=sendonly)){
 set_dlg_flag(7);
   } else {
   if ( search_body(a=inactive.)){
  set_dlg_flag(8);
  }
}
if (is_dlg_flag_set(7)  (status==200)){

replace_body(a=sendrecv.,a=recvonly);
} else {
if (is_dlg_flag_set(8)  
(status==200)){

replace_body(a=sendrecv.,a=inactive);
}
}   
} else {
   if (is_method(INVITE)){
reset_dlg_flag(7);
reset_dlg_flag(8);
  }
} 
 }

onreply_route[2] {
if (is_dlg_flag_set(7)  (status==200)){
replace_body(a=sendrecv.,a=recvonly);
  } else {
if (is_dlg_flag_set(8)  (status==200)){

replace_body(a=sendrecv.,a=inactive);
}
}   
}

*



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Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

[spady]

Hi Binan, seems your idea does not work. I post ngrep traces. Look at last
200OK from IP 172.16.55.100 (opensips) to IP 10.9.101.163 (lync). SDP still
have a=inactive.

Snippet of code:

branch_route[2] {
if (is_method(INVITE)  is_audio_on_hold()) {
  xlog( L_ERR, LOG: Setto la flag \n );
 set_dlg_flag(7);
  }
  
if (is_dlg_flag_set(7)  (status==200)) {
xlog( L_ERR, LOG: SDP-ON-HOLD Sostituzione del 
parametro
INACTIVE\n );
   replace_body(a=sendrecv., a=inactive);
  }   

xlog(new branch at $ru\n);
}



T 2012/10/10 11:45:56.587382 10.9.101.163:61892 - 172.16.55.100:5060 [AP]
INVITE sip:172.16.52.51 SIP/2.0.
FROM: sip:3707@172.16.55.100;user=phone;epid=D430E933C4;tag=198e999071.
TO: sip:172.16.52.51;tag=8b30fc41e5be9ba91c2d8e1b152a7e9a.
CSEQ: 2 INVITE.
CALL-ID: 8fc28eebc1aa2e142dd34dd6d9f4713d@172.16.52.51.
MAX-FORWARDS: 70.
VIA: SIP/2.0/TCP 10.9.101.163:61892;branch=z9hG4bKa8eef17.
ROUTE:
sip:172.16.55.100;transport=tcp;r2=on;lr,sip:172.16.55.100;r2=on;lr.
CONTACT:
sip:LYNC.lwtec.eu:5068;transport=Tcp;maddr=10.9.101.163;ms-opaque=dcf7ba7515f99f91.
CONTENT-LENGTH: 255.
SUPPORTED: timer.
SUPPORTED: 100rel.
USER-AGENT: RTCC/4.0.0.0 MediationServer.
CONTENT-TYPE: application/sdp.
Session-Expires: 1800.
Min-SE: 90.
.
v=0.
o=- 115 3 IN IP4 10.9.101.163.
s=session.
c=IN IP4 10.9.101.163.
b=CT:1000.
t=0 0.
m=audio 56346 RTP/AVP 8 106.
c=IN IP4 10.9.101.163.
a=rtcp:56347.
a=label:Audio.
a=sendrecv.
a=rtpmap:8 PCMA/8000.
a=rtpmap:106 telephone-event/8000.
a=fmtp:106 0-16.


T 2012/10/10 11:45:56.591138 172.16.55.100:5060 - 10.9.101.163:61892 [AP]
SIP/2.0 100 Giving a try.
FROM: sip:3707@172.16.55.100;user=phone;epid=D430E933C4;tag=198e999071.
TO: sip:172.16.52.51;tag=8b30fc41e5be9ba91c2d8e1b152a7e9a.
CSEQ: 2 INVITE.
CALL-ID: 8fc28eebc1aa2e142dd34dd6d9f4713d@172.16.52.51.
VIA: SIP/2.0/TCP 10.9.101.163:61892;branch=z9hG4bKa8eef17.
Server: OpenSIPS-Longwave.
Content-Length: 0.
.


U 2012/10/10 11:45:56.591494 172.16.55.100:5060 - 172.16.52.51:5060
INVITE sip:172.16.52.51 SIP/2.0.
Record-Route: sip:172.16.55.100;r2=on;lr.
Record-Route: sip:172.16.55.100;transport=tcp;r2=on;lr.
FROM: sip:3707@172.16.55.100;user=phone;epid=D430E933C4;tag=198e999071.
TO: sip:172.16.52.51;tag=8b30fc41e5be9ba91c2d8e1b152a7e9a.
CSEQ: 2 INVITE.
CALL-ID: 8fc28eebc1aa2e142dd34dd6d9f4713d@172.16.52.51.
MAX-FORWARDS: 69.
Via: SIP/2.0/UDP 172.16.55.100;branch=z9hG4bKf65f.5b426246.0;i=5.
VIA: SIP/2.0/TCP 10.9.101.163:61892;branch=z9hG4bKa8eef17.
CONTACT:
sip:LYNC.lwtec.eu:5068;transport=Tcp;maddr=10.9.101.163;ms-opaque=dcf7ba7515f99f91.
CONTENT-LENGTH: 255.
SUPPORTED: timer.
SUPPORTED: 100rel.
USER-AGENT: RTCC/4.0.0.0 MediationServer.
CONTENT-TYPE: application/sdp.
Session-Expires: 1800.
Min-SE: 90.
.
v=0.
o=- 115 3 IN IP4 10.9.101.163.
s=session.
c=IN IP4 10.9.101.163.
b=CT:1000.
t=0 0.
m=audio 56346 RTP/AVP 8 106.
c=IN IP4 10.9.101.163.
a=rtcp:56347.
a=label:Audio.
a=sendrecv.
a=rtpmap:8 PCMA/8000.
a=rtpmap:106 telephone-event/8000.
a=fmtp:106 0-16.


U 2012/10/10 11:45:56.604032 172.16.52.51:5060 - 172.16.55.100:5060
SIP/2.0 100 Trying.
To: sip:172.16.52.51;tag=8b30fc41e5be9ba91c2d8e1b152a7e9a.
From: sip:3707@172.16.55.100;user=phone;tag=198e999071;epid=D430E933C4.
Call-ID: 8fc28eebc1aa2e142dd34dd6d9f4713d@172.16.52.51.
CSeq: 2 INVITE.
Via: SIP/2.0/UDP 172.16.55.100;branch=z9hG4bKf65f.5b426246.0;i=5.
Via: SIP/2.0/TCP 10.9.101.163:61892;branch=z9hG4bKa8eef17.
Content-Length: 0.
.


U 2012/10/10 11:45:56.626543 172.16.52.51:5060 - 172.16.55.100:5060
SIP/2.0 200 OK.
Record-Route: sip:172.16.55.100;r2=on;lr.
Record-Route: sip:172.16.55.100;transport=tcp;r2=on;lr.
Content-Type: application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE.
Contact: Pippo sip:172.16.52.51.
Require: timer.
Supported: 100rel,timer,from-change.
User-Agent: OXO_GW_820/044.001.
Session-Expires: 1800;refresher=uac.
P-Asserted-Identity: Pippo sip:172.16.52.51.
To: sip:172.16.52.51;tag=8b30fc41e5be9ba91c2d8e1b152a7e9a.
From: sip:3707@172.16.55.100;user=phone;tag=198e999071;epid=D430E933C4.
Call-ID: 8fc28eebc1aa2e142dd34dd6d9f4713d@172.16.52.51.
CSeq: 2 INVITE.
Via: SIP/2.0/UDP 172.16.55.100;branch=z9hG4bKf65f.5b426246.0;i=5.
Via: SIP/2.0/TCP 10.9.101.163:61892;branch=z9hG4bKa8eef17.
Content-Length: 219.
.
v=0.
o=default 1349862354 1349862356 IN IP4 172.16.52.51.
s=session.
c=IN IP4 172.16.52.51.
t=0 0.
m=audio 32000 RTP/AVP 8 106.
a=sendrecv.
a=rtpmap:106 telephone-event/8000.
a=fmtp:106 0-15.
a=ptime:30.
a=maxptime:90.


*T 2012/10/10 11:45:56.627802 172.16.55.100:5060 - 10.9.101.163:61892 [AP]*
SIP/2.0 200 OK.
Record-Route: sip:172.16.55.100;r2=on;lr.
Record-Route: sip:172.16.55.100;transport=tcp;r2=on;lr.
Content-Type: application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE.
Contact: Pippo sip:172.16.52.51.
Require: timer.
Supported: 100rel,timer,from-change.
User-Agent: OXO_GW_820/044.001.

Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

[spady]

SOLVED!!
i was not reseted flag!!! :-( 

This is the right code:

branch_route[2] {
if (is_method(INVITE)  is_audio_on_hold()) {
 
 set_dlg_flag(7);
  } else {
   if (is_method(INVITE)){
reset_dlg_flag(7);
  }
 } 
if (is_dlg_flag_set(7)  (status==200)) {

   replace_body(a=sendrecv., a=inactive);
  }   

xlog(new branch at $ru\n);
  }




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Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

[spady]

I talked to early !! :-(
It's happening  a very strange thing.
Seems that this part of code is no more taken into account:

*if (is_method(INVITE) || is_audio_on_hold()){
 set_dlg_flag(7);
 
  } else {
   if (is_method(INVITE)){
reset_dlg_flag(7);
  }
*

I mean the part relating to is_audio_on_hold() infact, flag is no more
reseted.
What can be changed? I can't understand!! 
What is the condition for is_audio_on_hold() to work? I mean, the function
is_audio_on_hold() which part of incoming SIP msg it looks for??

Thanks



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Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

[spady]

Hi Binan,
as I said, i tried both methods but seems nothing is changing.
How can I test if seted flag is really seted? How can i show it on logs??




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[OpenSIPS-Users] dlg_validate_dialog Error, help

[spady]

Hi, while i was making some test for my lab ( place on hold and retrive it )
i noticed on opensips's log the following error:

*ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
dlg=[sip:5100@172.16.52.51;transport=UDP;user=phone] ,
req=[sip:5100@172.16.52.51;user=phone] *

I cannot understand why i get it. 
There someone that can explain why? what's wrong?

here is the entire SIP log of call.
Thanks a lot



U 2012/10/10 17:38:13.909354 172.16.52.51:5060 - 172.16.55.100:5060
INVITE sip:3707@172.16.55.100;user=phone SIP/2.0.
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE.
Supported: 100rel,from-change,timer,histinfo.
User-Agent: OXO_GW_820/044.001.
Session-Expires: 43200.
P-Asserted-Identity: Pippo sip:5100@172.16.52.51;user=phone.
History-Info: sip:3707@172.16.55.100;user=phone;index=1.
To: sip:3707@172.16.55.100;user=phone.
From: Pippo
sip:5100@172.16.52.51;user=phone;tag=13ac8df865d64dcd009847fc0a655633.
Contact: Pippo sip:5100@172.16.52.51;transport=UDP;user=phone.
Content-Type: application/sdp.
Call-ID: 3344b6afae639b1b0e2b1ce2b9ace640@172.16.52.51.
CSeq: 2006862861 INVITE.
Via: SIP/2.0/UDP
172.16.52.51;rport;branch=z9hG4bKd5d0ea77f8984e6c78d00a415ad7d86e.
Max-Forwards: 70.
Content-Length: 215.
.
v=0.
o=default 1349883523 1349883523 IN IP4 172.16.52.51.
s=-.
c=IN IP4 172.16.52.51.
t=0 0.
m=audio 32000 RTP/AVP 8 106 0.
a=sendrecv.
a=rtpmap:106 telephone-event/8000.
a=fmtp:106 0-15.
a=ptime:30.
a=maxptime:90.


U 2012/10/10 17:38:13.912215 172.16.55.100:5060 - 172.16.52.51:5060
SIP/2.0 100 Giving a try.
To: sip:3707@172.16.55.100;user=phone.
From: Pippo
sip:5100@172.16.52.51;user=phone;tag=13ac8df865d64dcd009847fc0a655633.
Call-ID: 3344b6afae639b1b0e2b1ce2b9ace640@172.16.52.51.
CSeq: 2006862861 INVITE.
Via: SIP/2.0/UDP
172.16.52.51;received=172.16.52.51;rport=5060;branch=z9hG4bKd5d0ea77f8984e6c78d00a415ad7d86e.
Server: OpenSIPS-Longwave.
Content-Length: 0.
.


T 2012/10/10 17:38:13.913014 172.16.55.100:53432 - 10.9.101.163:5068 [AP]
INVITE sip:3...@lync.lwtec.eu:5068;user=phone SIP/2.0.
Record-Route: sip:172.16.55.100;transport=tcp;r2=on;lr;did=9b9.e8f0d7d7.
Record-Route: sip:172.16.55.100;r2=on;lr;did=9b9.e8f0d7d7.
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE.
Supported: 100rel,from-change,timer,histinfo.
User-Agent: OXO_GW_820/044.001.
Session-Expires: 43200.
P-Asserted-Identity: Pippo sip:5100@172.16.52.51;user=phone.
History-Info: sip:3707@172.16.55.100;user=phone;index=1.
To: sip:3707@172.16.55.100;user=phone.
From: Pippo
sip:5100@172.16.52.51;user=phone;tag=13ac8df865d64dcd009847fc0a655633.
Content-Type: application/sdp.
Call-ID: 3344b6afae639b1b0e2b1ce2b9ace640@172.16.52.51.
CSeq: 2006862861 INVITE.
Via: SIP/2.0/TCP 172.16.55.100;branch=z9hG4bKe686.f7ef5b43.0.
Via: SIP/2.0/UDP
172.16.52.51;received=172.16.52.51;rport=5060;branch=z9hG4bKd5d0ea77f8984e6c78d00a415ad7d86e.
Max-Forwards: 69.
Content-Length: 215.
.
v=0.
o=default 1349883523 1349883523 IN IP4 172.16.52.51.
s=-.
c=IN IP4 172.16.52.51.
t=0 0.
m=audio 32000 RTP/AVP 8 106 0.
a=sendrecv.
a=rtpmap:106 telephone-event/8000.
a=fmtp:106 0-15.
a=ptime:30.
a=maxptime:90.


T 2012/10/10 17:38:13.916062 10.9.101.163:5068 - 172.16.55.100:53432 [AP]
SIP/2.0 100 Trying.
FROM:
Pipposip:5100@172.16.52.51;user=phone;tag=13ac8df865d64dcd009847fc0a655633.
TO: sip:3707@172.16.55.100;user=phone.
CSEQ: 2006862861 INVITE.
CALL-ID: 3344b6afae639b1b0e2b1ce2b9ace640@172.16.52.51.
VIA: SIP/2.0/TCP 172.16.55.100;branch=z9hG4bKe686.f7ef5b43.0,SIP/2.0/UDP
172.16.52.51;received=172.16.52.51;branch=z9hG4bKd5d0ea77f8984e6c78d00a415ad7d86e;rport=5060.
CONTENT-LENGTH: 0.
.


T 2012/10/10 17:38:14.682343 10.9.101.163:5068 - 172.16.55.100:53432 [AP]
SIP/2.0 183 Session Progress.
FROM:
Pipposip:5100@172.16.52.51;user=phone;tag=13ac8df865d64dcd009847fc0a655633.
TO: sip:3707@172.16.55.100;user=phone;tag=79c7c2c84d;epid=D430E933C4.
CSEQ: 2006862861 INVITE.
CALL-ID: 3344b6afae639b1b0e2b1ce2b9ace640@172.16.52.51.
VIA: SIP/2.0/TCP 172.16.55.100;branch=z9hG4bKe686.f7ef5b43.0,SIP/2.0/UDP
172.16.52.51;received=172.16.52.51;branch=z9hG4bKd5d0ea77f8984e6c78d00a415ad7d86e;rport=5060.
RECORD-ROUTE:
sip:172.16.55.100;transport=tcp;r2=on;lr;did=9b9.e8f0d7d7,sip:172.16.55.100;r2=on;lr;did=9b9.e8f0d7d7.
CONTACT: sip:LYNC.lwtec.eu:5068;transport=Tcp;maddr=10.9.101.163.
CONTENT-LENGTH: 255.
CONTENT-TYPE: application/sdp.
ALLOW: CANCEL.
ALLOW: BYE.
ALLOW: UPDATE.
ALLOW: PRACK.
REQUIRE: 100rel.
SERVER: RTCC/4.0.0.0 MediationServer.
Rseq: 1.
.
v=0.
o=- 219 1 IN IP4 10.9.101.163.
s=session.
c=IN IP4 10.9.101.163.
b=CT:1000.
t=0 0.
m=audio 55904 RTP/AVP 8 106.
c=IN IP4 10.9.101.163.
a=rtcp:55905.
a=label:Audio.
a=sendrecv.
a=rtpmap:8 PCMA/8000.
a=rtpmap:106 telephone-event/8000.
a=fmtp:106 0-16.


U 2012/10/10 17:38:14.683662 172.16.55.100:5060 - 172.16.52.51:5060
SIP/2.0 183 Session Progress.
FROM:
Pipposip:5100@172.16.52.51;user=phone;tag=13ac8df865d64dcd009847fc0a655633.
TO: 

Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

[spady]

Any idea?



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[OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

[spady]

Hi all, i am trying to realize a TCP/UDP gateway, using Opensips 1.8.1, to
connect a IPPBX to Lync Server 2010.

IPPBX---(Trunk SIP)---OPENSIPS---(Trunk SIP)---LYNC

Ip pbx does not support TCP protocol and Lync does support only TCP and TLS
so i need a proxy to do this job.
For now calls can go in both direction but problem comes when from Lync side
i try to put on hold the call.
Lync, for putting on hold, send a new INVITE with a=inactive in SDP body. IP
Pbx does not support/understand this parameter and call can't go on hold. Ip
PBX replies to this INVITE whit a 200 OK with a=sendrecv in SDP.
What I thought is using  if (status== 200)   then rewrite SDP body from
a=sendrecv ( PBX Side ) to a=inactive so that Lync can understand it and
call goes on hold. 
The problem is when Lync user wants to retrive call. Lync send again another
INVITE, with a=sendrecv, IP PBX then reply with 200 OK and a=sendrecv BUT (
here is the problem ) Opensips code ( that i used before ) rewrite again 200
OK message changing a=sendrecv to a=inactive and RTP flow does not start (
as aspected ).
Does anyone have some good idea to solve this problem? Any ideas?
Hope I explained in a clear manner :-)

Thanks to all



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Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

[spady]

Hi Binan, thanks for reply.
I understood your idea and I am going to try but i don't know how to create
a condition. let me explain:

I could do something like:
if (is_method(INVITE)) ...
... set_dlg_flag(3)
..

but how to tell to Opensips to check a=inactive / a=sendrecv ??? 
if((a=inactive))

Thanks



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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

Hi Binan,
don't care about it. I made a lot of changing to figure out the issue so
traces could be different to one another. Configuration of opensips and
rtpproxy are always correct ( I mean the two part was always matched ).
Thanks anyway :-)





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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

I have some news but issue still present.
I used unix socket, instead of udp, but RTPPROXY crashes at each call.
Below traces of opensips side and rtpproxy side:

OPENSIPS:

*Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26026]:
ERROR:rtpproxy:send_rtpp_command: can't read reply from a RTP proxy
Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26026]:
ERROR:rtpproxy:send_rtpp_command: proxy unix:/var/run/rtpproxy.sock does
not respond, disable it
Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26026]:
ERROR:rtpproxy:force_rtp_proxy_body: no available proxies
Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26020]:
ERROR:rtpproxy:force_rtp_proxy: Unable to parse body
Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26021]:
ERROR:rtpproxy:force_rtp_proxy: Unable to parse body
Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26024]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies
Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26022]:
ERROR:rtpproxy:force_rtp_proxy: Unable to parse body
Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26022]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies
Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26022]:
ERROR:rtpproxy:engage_close_callback: cannot unforce rtp proxy
Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26025]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies
Sep 27 13:02:03 opensips /usr/local/opensips_proxy/sbin/opensips[26026]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies
Sep 27 13:02:05 opensips /usr/local/opensips_proxy/sbin/opensips[26024]:
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies *


RTPPROXY:

*Sep 27 13:01:46 opensips rtpproxy[25993]: DBUG:handle_command: received
command VF 20071116
Sep 27 13:01:57 opensips rtpproxy[25993]: DBUG:handle_command: received
command UIER172.16.52.121c0,8,18,101
ZGY2Yjk5ZGJiNDQ0ZWI0MGVhZmMzODAxNDc3YzgwMjI 151.x.x.19 60242 671d9c3d;1


  
*

Any idea???




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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

I tried but i get the following error:

*Sep 27 15:09:49 opensips opensips: ERROR:core:set_mod_param_regex:
parameter rtpproxy_sock not found in module nathelper*

Seems that parameter you suggested me is not more avaible in nathelper
module.

I started rtpproxy like below

*root 27019 1  0 15:05 ?00:00:00 rtpproxy -F -s
udp:127.0.0.1 7890 10.9.23.41/151.8.12.201 -u root -d DBUG LOG_LOCAL2*





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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

I tried to use RTPPROXY_CLIENT but i get an error:

*root@opensips:/tmp# java rtpproxy-client-api-0.2.jar
Exception in thread main java.lang.NoClassDefFoundError:
rtpproxy-client-api-0/2/jar
Caused by: java.lang.ClassNotFoundException: rtpproxy-client-api-0.2.jar
at java.net.URLClassLoader$1.run(URLClassLoader.java:202)
at java.security.AccessController.doPrivileged(Native Method)
at java.net.URLClassLoader.findClass(URLClassLoader.java:190)
at java.lang.ClassLoader.loadClass(ClassLoader.java:306)
at sun.misc.Launcher$AppClassLoader.loadClass(Launcher.java:301)
at java.lang.ClassLoader.loadClass(ClassLoader.java:247)
Could not find the main class: rtpproxy-client-api-0.2.jar.  Program will
exit.*

What that means? I suppose some trouble with java but i cannot understand
what.




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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

Yes, it is!
I already tried with different users, both opensips and rtpproxy.




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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

Hi Max,
*No available proxies* and so on is because after first error ( opensips
can't connect to rtpproxy and then it disable rtpproxy momentaly ).

btw the port is listening:

*root@opensips:/tmp# netstat -anp|grep rtpproxy
udp0  0 127.0.0.1:7890  0.0.0.0:*  
27019/rtpproxy
unix  2  [ ] DGRAM5755127019/rtpproxy*





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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

Hi Binan,
i checked the debian and i really think that SELinux is not implemented and
is obviuosly NOT RUNNING!!

Maybe i did not understand you. Should be running?




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[OpenSIPS-Users] Rtpproxy connection

[spady]

Hi all, I thought to open a new threat about my issue because is not ( i
suppose ) related to scripting.

My issue is that any time rtpproxy has to be invoked by opensips script i
get the following error:

*ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
Connection refused*

and rtpproxy shuts down.
I am investigating on it by some weeks and seems to be an hard problem ( for
me :-) ) .
What I am asking is in which way OPenSips try to connect to rtpproxy. What's
the command that it tries to send but it can't because Connection is
refused. Who refused the connection? 
Can someone explain me all of that so i can figured out my original problem.
Thanks to all




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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

Hi Binan, thanks for your reply.
I read links you posted me but i cannot understand how to send command to
rtpproxy to test it.

I tried from CLI like:

*root@opensips:~# rtpproxy VF 20050322*

but i get no response. Probably is not the way like it should. Can you
explain me better?
Thanks again.




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Re: [OpenSIPS-Users] Rtpproxy connection

[spady]

- No!, after opensips module sends command to rtpproxy this one shut down ( i
cannot see anymore with ps command )

- in rtpproxy logs i cannot see any specific error. Below example of code
that I get from rtpproxy:
*Sep 27 00:31:01 opensips rtpproxy[19991]: DBUG:handle_command: received
command 20126_5 UIER172.16.52.121c0,8,18,101
M2EzYzNmMmRmYzg0MGNiY2M1YjhjMTUzMWFkOWNiYTg 37.103.117.107 60234 cfd2be4f;1
Sep 27 00:31:01 opensips rtpproxy[19991]: INFO:handle_command: new session
M2EzYzNmMmRmYzg0MGNiY2M1YjhjMTUzMWFkOWNiYTg, tag cfd2be4f;1 requested, type
strong*

As you can see, no error!!!

P.S. I started rtpproxy like below:
*rtpproxy -F -s udp:127.0.0.1:10177 -l 10.9.23.41/151.x.x.201 -u root -d
DBUG:LOG_LOCAL2*



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Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

[spady]

Hi again and sorry for big delay but i was out of office for long time.
Today I started to have some other test and always get same issue!!! I also
tried with a new installation ( debian+opensips 1.8.1+new IPs+rtpproxy). As
Sam suggested a tried using engage_rtpproxy but always same issue.
I really think problem is just before that script is taken into account.
Errors says that opensips CAN'T send command to rtpproxy because connection
 is refused!!.
How can I test manually connection to RTPPROXY? Is there such a command from
CLI that i can try?
Hope someone can help me.
Thanks.




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Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

[spady]

Hi,
I have a news regarding logs.
Under /var/log/messages I found this error:


Sep  7 12:52:52 opensips kernel: [584059.568944] rtpproxy[2256]: segfault at
0 ip 08051334 sp b6ce60f0 error 4 in rtpproxy[8048000+e000]

Any idea about what that means??



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Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

[spady]

I installed it by

git clone git://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy 

then ./configure and then make, make install.

Sorry if i cannot understand but i'am not so skilled with opensips.
I believe that code, where rtpproxy is called, is this:

route[1] { 
# for INVITEs enable some additional helper routes 
if (is_method(INVITE)) { 

if (isflagset(10)) { 
rtpproxy_offer(ro); 
} 

t_on_branch(2); 
t_on_reply(2); 
t_on_failure(1); 
} 

if (isflagset(10)) { 
add_rr_param(;nat=yes); 
} 

if (!t_relay()) { 
send_reply(500,Internal Error); 
}; 
exit; 
} 
branch_route[2] { 
xlog(new branch at $ru\n); 
} 
onreply_route[2] { 
if (nat_uac_test(1)) 
fix_nated_contact(); 
if ( isflagset(10) ) 
rtpproxy_answer(ro); 
xlog(incoming reply\n); 
}  



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Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

[spady]

Hi, can someone help me?



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Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

[spady]

Hi Bogdan,
NO, after made call rtpproxy is no more running!!!



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Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

[spady]

Hi Sam,
i've not created a /etc/init.d/rtpproxy file!! Have I to create manually or
is inside rtpproxy's installation dir?
Btw, as wrote above, I start RTPP. by using this command:

rtpproxy -F -s udp:127.0.0.1:7890 -l 10.9.23.41/151.x.x.201 -d
DBUG:LOG_LOCAL2 

and also Yes, it crash when i make a call!!

Any idea?





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Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

[spady]

Hi, I am making some test and also changing configuration at opensips.cfg, i
have always same issue.

The problem is related to this ERROR:

ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
Connection refused

But i cannot understand why RTPPROXY refuses connection.
Any idea?




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[OpenSIPS-Users] RTPPROXY shutdown at each call

[spady]

Hi all, I am implementing my first OpenSIPS with RTPPROXY so, for sure, any
issue is related to my NOT-Knowledge!! :-)

Issue is follow:
each time I make a call ( from USERA to USERB, for instance ) RTPPROXY shuts
down immediatly. 
Here there are logs and config:


STARTING SERVER:

Sep  4 16:12:20 opensips opensips: DBG:core:yyparse: loading module
/usr/local/opensips_proxy/lib/opensips/modules/rtpproxy.so
Sep  4 16:12:20 opensips opensips: DBG:core:set_mod_param_regex: rtpproxy
matches module rtpproxy
Sep  4 16:12:20 opensips opensips: DBG:core:set_mod_param_regex: found
rtpproxy_sock in module rtpproxy
[/usr/local/opensips_proxy/lib/opensips/modules/]
Sep  4 16:12:20 opensips opensips: DBG:core:set_mod_param_regex: rtpproxy
matches module rtpproxy
Sep  4 16:12:20 opensips opensips: DBG:core:set_mod_param_regex: found
rtpproxy_autobridge in module rtpproxy
[/usr/local/opensips_proxy/lib/opensips/modules/]
Sep  4 16:12:20 opensips opensips: DBG:core:find_cmd_export_t: found
rtpproxy_offer(1) in module rtpproxy
[/usr/local/opensips_proxy/lib/opensips/modules/]
Sep  4 16:12:20 opensips opensips: DBG:core:find_cmd_export_t: found
rtpproxy_answer(1) in module rtpproxy
[/usr/local/opensips_proxy/lib/opensips/modules/]
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19707]:
DBG:core:init_mod: initializing module rtpproxy
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19707]:
DBG:rtpproxy:add_rtpproxy_socks: url is udp:127.0.0.1:7890, len is 18
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19707]:
DBG:core:init_mod: register MI for rtpproxy
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19707]:
DBG:core:fix_actions: fixing rtpproxy_offer, line 486
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19707]:
DBG:core:fix_actions: fixing rtpproxy_answer, line 516
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19710]:
DBG:core:init_mod_child: type=CHILD, rank=2, module=rtpproxy
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19710]:
DBG:rtpproxy:connect_rtpproxies: [RTPProxy] set list 0xaee30b0c
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19710]:
DBG:rtpproxy:connect_rtpproxies: [Re]connecting sockets (1  0)
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19710]:
DBG:rtpproxy:connect_rtpproxies: connected 127.0.0.1:7890
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19708]:
DBG:core:init_mod_child: type=PROC_MODULE, rank=-2, module=rtpproxy
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19711]:
DBG:core:init_mod_child: type=CHILD, rank=3, module=rtpproxy
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19711]:
DBG:rtpproxy:connect_rtpproxies: [RTPProxy] set list 0xaee30b0c
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19711]:
DBG:rtpproxy:connect_rtpproxies: [Re]connecting sockets (1  0)
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19711]:
DBG:rtpproxy:connect_rtpproxies: connected 127.0.0.1:7890
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19711]:
INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for
it enabled
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19711]:
DBG:rtpproxy:raise_rtpproxy_event: no event sent
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19708]:
DBG:rtpproxy:connect_rtpproxies: [RTPProxy] set list 0xaee30b0c
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19708]:
DBG:rtpproxy:connect_rtpproxies: [Re]connecting sockets (1  0)
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19708]:
DBG:rtpproxy:connect_rtpproxies: connected 127.0.0.1:7890
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19708]:
INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for
it enabled
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19708]:
DBG:rtpproxy:raise_rtpproxy_event: no event sent
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19712]:
DBG:core:init_mod_child: type=CHILD, rank=4, module=rtpproxy
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19712]:
DBG:rtpproxy:connect_rtpproxies: [RTPProxy] set list 0xaee30b0c
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19712]:
DBG:rtpproxy:connect_rtpproxies: [Re]connecting sockets (1  0)
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19712]:
DBG:rtpproxy:connect_rtpproxies: connected 127.0.0.1:7890
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19712]:
INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for
it enabled
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19712]:
DBG:rtpproxy:raise_rtpproxy_event: no event sent
Sep  4 16:12:20 opensips /usr/local/opensips_proxy/sbin/opensips[19710]:
INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 

[OpenSIPS-Users] External SIP require AUTH

[spady]

Hi all, maybe is a dummy question but I don't know how to achieve solution.
I am connecting a External SIP LINE provided by a Pubblic ISP.
It require authentication for each incoming call ( outgoing call by opensips
point of view ).
How can i solve it? Is there a module or a db table where store credentials
and then use them by a snippet of code??
Help,
thanks




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Re: [OpenSIPS-Users] Add or modify grp table

[spady]

Thanks Bogdan,
that was the solution!!

Best regards.




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[OpenSIPS-Users] Add or modify grp table

[spady]

Hi all,
by default ( i thought ) there are 3 groups that i can use whit grp table:
local,int and ld.
I would like to use my own ( i.e: nazionali,locali and so on.. )
Is there a way to add them to database?
Now, if a try command 

./opensipsctl acl grant lifesize1@172.16.55.100 nazionali

i get this error:

WARNING: Invalid privilege: acl 'nazionali' ignored

Thanks




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Re: [OpenSIPS-Users] Forward NOTIFY msg,how to do that?

[spady]

Yes, very similar to mine. 
Tell me if you will be able to achive your target.
Btw, using google, there are some good blogs that they explain integration
of opensips into asterisk box very well.
If you need help, ask me. No problem.
Have a nice day


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Re: [OpenSIPS-Users] Forward NOTIFY msg,how to do that?

[spady]

Hi Chris,
for the moment I am non working on this case. I  am working with a fax
server and pbx-ip, placing opensips in the middle.
Btw, the code you can see on this thread should work. 
Where are you stuck?

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[OpenSIPS-Users] Query on external source ( execel or DB )

[spady]

Hi folk,
I need to know if with OpenSIPS i am able to have the following scenario:

INITIAL SCENARIO:

Incoming external call goes into SIP PBX , this one routes call toward
another SIP PBX.

WANTED SCENARIO:

Incoming external call goes into SIP PBX,this one routes call toward
OpeSIPS ( OpenSIPS have to check into outside source, for example a execl
file or access file if calling number is present. If true It has to modify
TO: header in a certain manner; if false, it has to modify TO Header with
another way ). After modification, OpenSIPS routes call into final SIP
PBX.

Is that passible? I know OpenSIPS can easly modify Sip headers but what I
don't know is if OpenSIPS can use a execl or acces external file as source
to get the condition ( true or false ).

Thanks


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Re: [OpenSIPS-Users] Query on external source ( execel or DB )

[spady]

Thanks Vlad for your reply.
Query has to be done into AS400's db. I think is not a standard db. So I
thought about excel or access file ( I can export data from AS400 ).
If there is not another solution, I will have a try as you suggested me.
Thnaks a lot.


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Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

[spady]

Hi, Can someone explain/help me on this 
I really will appreciate 
Best Regards

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Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

[spady]

Hi Denis, I know. it's wanted.
Is changed to 87019.



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Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

[spady]

Hi Denis, that was it!!! It was setted to auto . 
I set it to none and now it works as aspected Perfet.
Thank you very much for your hint ;-)
Best regards

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Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

[spady]

Can someone help me with this?

I checked again config and seems ok but form CP nothing yet.

Regards

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[OpenSIPS-Users] how does OpenSIPS manage 183's message??

[spady]

Hi all, I am still testing my solution to provide some additional features to
own Fax server, thanks to OpenSIPS.

IP-PBX OpenSIPS  Fax Server

I am using OpenSIPS in stateless mode ( so without record-route ) and this
is the sip trace at OpenSIPS level

U 2011/12/13 15:46:04.075195 172.16.52.7:5061 - 10.9.101.166:5060
INVITE sip:0363394686180@10.9.101.166:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 172.16.52.7:5061.
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=11D255FDC152.
To: sip:0363394686180@10.9.101.166:5060.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: sip:+390522375507@172.16.52.7:5061;user=phone.
User-Agent: Alcatel-Lucent OmniTouch Fax Server Application/6.5.6.28.
P-Alcatel-CSBU: charging=sip:2542@ucalcatel.sedoc.locale.
Content-Type: application/sdp.
Content-Length: 211.
.
v=0.
o=XMedius-Fax-Gateway 55439616 616 IN IP4 172.16.52.7.
s=SIP Fax Call.
c=IN IP4 172.16.52.7.
t=0 0.
m=audio 46022 RTP/AVP 18 8.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:8 PCMA/8000.
a=ptime:20.


U 2011/12/13 15:46:04.080484 10.9.101.166:5060 - 172.16.52.7:5061
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 172.16.52.7:5061.
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=11D255FDC152.
To: sip:0363394686180@10.9.101.166:5060.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Server: OpenSIPS-Longwave.
Content-Length: 0.
.


U 2011/12/13 15:46:04.080927 10.9.101.166:5060 - 10.9.6.3:5060
INVITE sip:87019363394686180@10.9.6.3:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.9.101.166;branch=z9hG4bK8106.0a3d7b63.0.
Via: SIP/2.0/UDP 172.16.52.7:5061.
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=11D255FDC152.
To: sip:87019363394686180@10.9.6.3:5060.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Max-Forwards: 69.
Contact: sip:+390522375507@172.16.52.7:5061;user=phone.
User-Agent: Alcatel-Lucent OmniTouch Fax Server Application/6.5.6.28.
P-Alcatel-CSBU: charging=sip:2542@ucalcatel.sedoc.locale.
Content-Type: application/sdp.
Content-Length: 211.
.
v=0.
o=XMedius-Fax-Gateway 55439616 616 IN IP4 172.16.52.7.
s=SIP Fax Call.
c=IN IP4 172.16.52.7.
t=0 0.
m=audio 46022 RTP/AVP 18 8.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:8 PCMA/8000.
a=ptime:20.


U 2011/12/13 15:46:04.082838 10.9.6.3:5060 - 10.9.101.166:5060
SIP/2.0 100 Trying.
To: sip:87019363394686180@10.9.6.3:5060.
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=11D255FDC152.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Via: SIP/2.0/UDP 10.9.101.166;branch=z9hG4bK8106.0a3d7b63.0.
Via: SIP/2.0/UDP 172.16.52.7:5061.
Content-Length: 0.
.


U 2011/12/13 15:46:04.506636 10.9.6.3:5060 - 10.9.101.166:5060
SIP/2.0 183 Session Progress.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE.
Contact: sip:10.9.6.3.
User-Agent: OmniPCX Enterprise R9.0 h1.301.50.
P-Alcatel-CSBU: categparty=external.
Content-Type: application/sdp.
To:
sip:87019363394686180@10.9.6.3:5060;tag=3439ea69c02c4dd7146a60c535fa4a06.
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=11D255FDC152.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Via: SIP/2.0/UDP 10.9.101.166;branch=z9hG4bK8106.0a3d7b63.0.
Via: SIP/2.0/UDP 172.16.52.7:5061.
Content-Length: 229.
.
v=0.
o=OXE 1323791155 1323791155 IN IP4 10.9.6.3.
s=abs.
c=IN IP4 10.9.6.8.
t=0 0.
m=audio 32560 RTP/AVP 18 96.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=ptime:30.
a=maxptime:40.
a=rtpmap:96 telephone-event/8000.
a=sendrecv.


U 2011/12/13 15:46:04.508542 10.9.101.166:5060 - 172.16.52.7:5061
SIP/2.0 183 Session Progress.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE.
Contact: sip:10.9.6.3.
User-Agent: OmniPCX Enterprise R9.0 h1.301.50.
P-Alcatel-CSBU: categparty=external.
Content-Type: application/sdp.
To:
sip:0363394686180@10.9.101.166:5060;tag=3439ea69c02c4dd7146a60c535fa4a06.
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=11D255FDC152.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Via: SIP/2.0/UDP 172.16.52.7:5061.
Content-Length: 229.
.
v=0.
o=OXE 1323791155 1323791155 IN IP4 10.9.6.3.
s=abs.
c=IN IP4 10.9.6.8.
t=0 0.
m=audio 32560 RTP/AVP 18 96.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=ptime:30.
a=maxptime:40.
a=rtpmap:96 telephone-event/8000.
a=sendrecv.


U 2011/12/13 15:46:06.028272 10.9.6.3:5060 - 10.9.101.166:5060
SIP/2.0 180 Ringing.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE.
Contact: sip:10.9.6.3.
User-Agent: OmniPCX Enterprise R9.0 h1.301.50.
P-Alcatel-CSBU: categparty=external.
Content-Type: application/sdp.
To:
sip:87019363394686180@10.9.6.3:5060;tag=3439ea69c02c4dd7146a60c535fa4a06.
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=11D255FDC152.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Via: SIP/2.0/UDP 

Re: [OpenSIPS-Users] Help with uac_replace_to

[spady]

Hi Vlad, thank you for your reply.

I found this solution and seems a good one. Can you confirm it?

if ($rU=~^0[0-9]+)
{
strip(1);
prefix(87019);
rewritehostport(10.9.6.3:5060);
uac_replace_to(sip:$rU@10.9.6.3:5060);
route(1);
exit;
}
 }  

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[OpenSIPS-Users] Help with uac_replace_to

[spady]

Hi all, I need to rewrite a To header, after send INVITE.
Here is my part of script:

if (avp_db_load($fu/username,$avp(ARS-OFS)))
 {
if (avp_check($avp(ARS-OFS),eq/lw-re/i))
{
if ($rU=~^0[0-9]+)
{
strip(1);
prefix(87019);
rewritehostport(10.9.6.3:5060);
uac_replace_to(sip:$ru);  
route(1);
exit;
}
 }
  else
  {
 if (avp_check($avp(ARS-OFS),eq/lw-mo/i))
 {
   if ($rU=~^0[0-9]+)
 {
   strip(1);
   prefix(87070);
   uac_replace_to(sip:$ru);
   route(1);
   exit;
 }
  }
}
}   

If I do like above, the result is:

INVITE sip:8701902700502403@10.9.6.3:5060;user=phone SIP/2.0.
Record-Route:
sip:10.9.101.166;lr;ftag=F542B897A1EA;vst=AENZQg0IBwUBCwQCBHABBR4LGgEDcR8BGA8UAx4FCjUwNjA7dXNlcj1waG9uZQ--.
Via: SIP/2.0/UDP 10.9.101.166;branch=z9hG4bK8ec5.cec86122.0.
Via: SIP/2.0/UDP 172.16.52.7:5061.
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=F542B897A1EA.
*To: sip:sip:8701902700502403@10.9.6.3:5060;user=phone.*
Call-ID: 13c1f9de-2f37-4856-b73b-634090ec4190@172.16.52.7.
CSeq: 101 INVITE.
Max-Forwards: 69.
Contact: sip:+390522375507@172.16.52.7:5061;user=phone.
User-Agent: Alcatel-Lucent OmniTouch Fax Server Application/6.5.6.28.
P-Alcatel-CSBU: charging=sip:2542@ucalcatel.sedoc.locale.
Content-Type: application/sdp.
Content-Length: 164.

But what I am trying to achive is something like this:

*To: sip:002700502403@10.9.6.3:5060*


How can I do that? 
Best regards.



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Re: [OpenSIPS-Users] Modify To:'s field and forward

[spady]

Hi list, I would share with you all what i reached about my script. Hope this
would be helpfull for someone of you.


if (avp_db_load($fu/username,$avp(ARS-OFS)))
 {
if (avp_check($avp(ARS-OFS),eq/lw-re/i))
{
if ($rU=~^0[0-9]+)
{
strip(1);
prefix(87019);
rewritehostport(10.9.6.3:5060);
route(1);
exit;
}
 }
  else
  {
 if (avp_check($avp(ARS-OFS),eq/lw-mo/i))
 {
   if ($rU=~^0[0-9]+)
 {
   strip(1);
   prefix(87070);
   rewritehostport(10.9.6.3:5060);
   route(1);
   exit;
 }
  }
  }
  else {
 rewritehostport(10.9.6.3:5060);
 route(1);
 exit;
}

Thank you all for your help.
Regards

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Re: [OpenSIPS-Users] Add Menu to OpenSIPS-CP

[spady]

Alex, just to be sure, Can I add more than one mysql's table at the time??
I tried to put also address tabel into local's file but seems it 's not
working.
Can you just confirm that?
Regards

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Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

[spady]

No one has idea?

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Re: [OpenSIPS-Users] Modify To:'s field and forward

[spady]

Hi Bogdan, thank you for your reply and your time.

I did as you sayd and it worked in part.
I had to modify this statement  *if ($rU=~^sip:0[0-9]{11}@) *   with *if
($rU=~^0[0-9]+)*

I don't really understood why. I really thought that regex 
*~^sip:0[0-9]{11}@*  was correct.

Anyway, thank you for your advise.


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Re: [OpenSIPS-Users] Modify To:'s field and forward

[spady]

I am trying to improve my script:

Now I would like that if avp_check fails, Opensips simply rewrite host and
port and keep ruri as recived.
I wrote the following but seems it's wrong 

if (avp_db_load($fu/username,$avp(Linea-LW)))
 {
xlog(L_DBG,
AVP_DB_LOAD Invocato!!\n);
if (avp_check($avp(Linea-LW),eq/lw/i))
{
if ($rU=~^0[0-9]+)
{
xlog(L_DBG,
URI modificato per invio verso OXE\n);
strip(1);
prefix(87019);
rewritehostport(10.9.6.3:5060);
route(1);
exit;
}
*else {
rewritehostport(10.9.6.3:5060);
route(1);
exit;*
}
}
}   

Can you drive me into right way?
Regards   

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Re: [OpenSIPS-Users] Modify To:'s field and forward

[spady]

Hi Again, i am tryng to do, as sayd, my first script.
I setted up the following:



 if (avp_db_load($fu/username,$avp(Linea-LW)))
{
xlog(L_DBG,
AVP_DB_LOAD Invocato!!\n);
if (avp_check($avp(Linea-LW),eq/y/i));
if ($rU=~^sip:0[0-9]{11}@)
{
xlog(L_DBG,
URI modificato per FAX SEVER\n);
strip(1);
prefix(87019);
rewritehostport(10.9.6.3:5060);
route(1);
exit;
}
}

But it does not work.

What's wrong?
Thanks in advance


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Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

[spady]

Yes, sure:


 $config-results_per_page = 25;
 $config-results_page_range = 10;

 // highlighting
 $config-from_color=black;  $config-from_bgcolor=yellow;
 $config-to_color=white;  $config-to_bgcolor=blue;
 $config-callid_color=black;  $config-callid_bgcolor=orange;
 $config-cseq_color=white;  $config-cseq_bgcolor=navy;
 $config-regexp_color=navy;   $config-regexp_bgcolor=red;

###

 //database tables
 $config-table_trace = sip_trace;

 $talk_to_this_assoc_id = 1 ;

 // sip proxy - ip:port

$proxy_list=array(udp:10.9.101.166:5060,tcp:10.9.101.166:5060,udp:127.0.0.1:5060,tcp:127.0.0.1:5060);
?


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Re: [OpenSIPS-Users] Add Menu to OpenSIPS-CP

[spady]

Thanksss Alex!!! It works perfectly!


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Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

[spady]

I done:


mysql SELECT DISTINCT callid FROM sip_trace WHERE status='' AND
direction='in' ORDER BY id DESC ;
Empty set (0.00 sec)

mysql
mysql



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Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

[spady]

Hi, i think it'l lastone:



opensips -V
version: opensips 1.7.1-notls (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM,
SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: unknown
@(#) $Id: main.c 8537 2011-11-08 17:02:11Z bogdan_iancu $
main.c compiled on 10:36:22 Nov 30 2011 with gcc 4.4.5


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Re: [OpenSIPS-Users] Modify To:'s field and forward

[spady]

Hi, i am gonna step by step closer but..it still not working.
This a log from opensips.log:



tail -f /var/log/opensips.log
Dec  5 16:42:06 opensips /sbin/opensips[2]: DBG:tm:timer_routine: timer
routine:2,tl=0xaf3edca0 next=(nil), timeout=237
Dec  5 16:42:06 opensips /sbin/opensips[2]: DBG:tm:wait_handler:
removing 0xaf3edc58 from table
Dec  5 16:42:06 opensips /sbin/opensips[2]: DBG:tm:delete_cell: delete
transaction 0xaf3edc58
Dec  5 16:42:06 opensips /sbin/opensips[2]: DBG:tm:wait_handler: done
Dec  5 16:42:08 opensips /sbin/opensips[22216]: DBG:core:udp_rcv_loop:
probing packet received len = 4
Dec  5 16:42:38 opensips /sbin/opensips[22217]: DBG:core:udp_rcv_loop:
probing packet received len = 4
Dec  5 16:43:01 opensips /sbin/opensips[22211]: DBG:mi_fifo:mi_parse_tree:
adding node  ; val all
Dec  5 16:43:01 opensips /sbin/opensips[22211]: DBG:mi_fifo:mi_parse_node:
end of input tree
Dec  5 16:43:01 opensips /sbin/opensips[22211]: DBG:mi_fifo:mi_fifo_server:
done parsing the mi tree
Dec  5 16:43:08 opensips /sbin/opensips[22216]: DBG:core:udp_rcv_loop:
probing packet received len = 4
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_msg: SIP
Request:
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_msg:  method: 
INVITE
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_msg:  uri:
sip:00522375568@10.9.101.166:5060;user=phone
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_msg: 
version: SIP/2.0
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_headers:
flags=2
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_via: end of
header reached, state=5
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_headers: via
found, flags=2
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_headers: this
is the first via
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:receive_msg: After
parse_msg...
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:receive_msg:
preparing to run routing scripts...
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:siptrace:sip_trace:
nothing to trace...
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_headers:
flags=100
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_to: end of
header reached, state=9
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_to:
display={}, ruri={sip:00522375568@10.9.101.166:5060}
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:get_hdr_field: To
[35]; uri=[sip:00522375568@10.9.101.166:5060]
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:get_hdr_field: to
body [sip:00522375568@10.9.101.166:5060#015#012]
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:get_hdr_field: cseq
CSeq: 101 INVITE
Dec  5 16:43:24 opensips /sbin/opensips[22217]:
DBG:maxfwd:is_maxfwd_present: value = 70
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:uri:has_totag: no totag
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_headers:
flags=78
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:tm:t_lookup_request:
start searching: hash=58302, isACK=0
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:tm:t_lookup_request:
proceeding to pre-RFC3261 transaction matching
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:tm:t_lookup_request: no
transaction found
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_to_param:
tag=B080AAD539C5
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_to: end of
header reached, state=29
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_to:
display={}, ruri={sip:+390522375507@172.16.52.7:5061}
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:grep_sock_info:
checking if host==us: 11==9   [172.16.52.7] == [127.0.0.1]
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:grep_sock_info:
checking if port 5060 matches port 5061
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:grep_sock_info:
checking if host==us: 11==12   [172.16.52.7] == [10.9.101.166]
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:grep_sock_info:
checking if port 5060 matches port 5061
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:grep_sock_info:
checking if host==us: 11==9   [172.16.52.7] == [127.0.0.1]
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:grep_sock_info:
checking if port 5060 matches port 5061
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:grep_sock_info:
checking if host==us: 11==12   [172.16.52.7] == [10.9.101.166]
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:grep_sock_info:
checking if port 5060 matches port 5061
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:check_self: host !=
me
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:parse_headers:
flags=200
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:get_hdr_field:
content_length=188
Dec  5 16:43:24 opensips /sbin/opensips[22217]: DBG:core:get_hdr_field:
found end of 

[OpenSIPS-Users] Add Menu to OpenSIPS-CP

[spady]

Hi List,
I was wondering is there is a way, and/or is possible, to add a menu to
OpenSIPS-CP under System Tab.
I would like to populate AVP Table by Control Panel. It would be easier then
do it by opensipsctl command.

Regards.

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Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

[spady]

No one has idea?

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Re: [OpenSIPS-Users] Modify To:'s field and forward

[spady]

Hi List, 
I searched into Forum and i am reading about AVP module and UAC module but
it's hard to write my first OpenSIPS's script :-(
As I sayd i am trying to modify the following INVITE message:

INVITE sip:00522375568@10.9.101.166:5060;user=phone SIP/2.0 
Via: SIP/2.0/UDP 172.16.52.7:5061 
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=B0A3C63723A2
To: sip:00522375568@10.9.101.166:5060 
Call-ID: 8f917371-ff97-4953-bc3d-fdba788fe45a@172.16.52.7 
CSeq: 101 INVITE 
Max-Forwards: 70 
Contact: sip:+390522375507@172.16.52.7:5061;user=phone
User-Agent: Alcatel-Lucent OmniTouch Fax Server Application/6.5.6.28 
P-Alcatel-CSBU: charging=sip:2542@ucalcatel.sedoc.locale 
Content-Type: application/sdp 
Content-Length: 235 

v=0 
o=XMedius-Fax-Gateway 79844629 629 IN IP4 172.16.52.7 
s=SIP Fax Call 
c=IN IP4 172.16.52.7 
t=0 0 
m=audio 62186 RTP/AVP 18 8 4 
a=rtpmap:18 G729/8000 
a=fmtp:18 annexb=no 
a=rtpmap:8 PCMA/8000 
a=rtpmap:4 G723/8000 
a=ptime:20

As Osiris sayd I started to write into usr_preferences table the following:

*opensipsctl avp add -T usr_preferences +390522375507@172.16.52.7 is_FAX 0
y*

+--+---+-+---+--+---+
| uuid | username  | domain  | attribute | type | value   |
+--+---+-+---+--+---+
|   | +390522375507 | 172.16.52.7 | is_FAX|0| y   |
+--+---+-+---+--+---+

then I should use AVP to delete first 0 from INVITE field and TO field and
change 10.9.101.166 (is OpenSIPS) with 10.9.6.3 (is PBX's ip)

INVITE sip:*0*0522375568@10.9.101.166:5060;user=phone SIP/2.0
and
To: sip:*0*0522375568@10.9.101.166:5060

and repleace them with 87019; so my new FROM and To will should became:

INVITE sip:*87019*0522375568@*10.9.6.3*:5060;user=phone SIP/2.0
and
To: sip:*87019*0522375568@*10.9.6.3*:5060

Please, can you help me to write this script/route logic 
I would really appreciate it.

Thanks in advance



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[OpenSIPS-Users] Add User From Opensips-Cp

[spady]

Hi all, I setted up a OpenSIPS-Cp and almost everything it's working good
except for User---User Managment-- Add New.
If I try to add a new User, when I press OK, nothing happen. User is not
added and I can't see anything into opensips.log.
I created some user by using opensipsctl add  and it works ( and I can see
created user form web interface ).
What's wrong?

I am using Opensips v. 1.7.1 and CP ver. 4.1

Best regards.


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Re: [OpenSIPS-Users] Add User From Opensips-Cp

[spady]

Hi Bogdan, thanks for reply. It was my mistake. I was creating a user,
whitout compiling all fields. ( i was missing aliases ).

Now it works perfectly.


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[OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

[spady]

Hi folk, i am facing this issue:
I set, just for learing purpose, my opensips.cfg as follow:

loadmodule siptrace.so

# - sip_trace --
modparam(siptrace, db_url,
mysql://opensips:opensipsrw@localhost/opensips)
modparam(siptrace, trace_on, 1)
modparam(siptrace, enable_ack_trace, 1)
modparam(siptrace, table, sip_trace)
modparam(siptrace, trace_flag, 22) 

### Routing Logic 


# main request routing logic

route{
sip_trace();
setflag(22);

Now if I make or receive a call i can see that siptrace is stored into mysql
db by this comand:

mysql select * from sip_trace ;

But on Control Panel there is nothing.
Here Apache2 logs when I press Show all

Thu Dec 01 23:42:12 2011] [error] [client 10.9.100.251] PHP Notice: 
Undefined variable: delete in
/var/www/opensips-cp/web/tools/system/siptrace/tracer.php on line 83,
referer:
http://10.9.101.166/cp/tools/system/siptrace/tracer.php?action=search
[Thu Dec 01 23:42:14 2011] [error] [client 10.9.100.251] File does not
exist: /var/www/favicon.ico 

and these if i press Search:

[Thu Dec 01 23:42:54 2011] [error] [client 10.9.100.251] PHP Notice: 
Undefined variable: delete in
/var/www/opensips-cp/web/tools/system/siptrace/tracer.php on line 83,
referer:
http://10.9.101.166/cp/tools/system/siptrace/tracer.php?action=search
[Thu Dec 01 23:42:54 2011] [error] [client 10.9.100.251] PHP Notice: 
Undefined variable: show_all in
/var/www/opensips-cp/web/tools/system/siptrace/tracer.php on line 89,
referer:
http://10.9.101.166/cp/tools/system/siptrace/tracer.php?action=search
[Thu Dec 01 23:42:54 2011] [error] [client 10.9.100.251] PHP Notice: 
Undefined variable: set_start in
/var/www/opensips-cp/web/tools/system/siptrace/tracer.php on line 104,
referer:
http://10.9.101.166/cp/tools/system/siptrace/tracer.php?action=search
[Thu Dec 01 23:42:54 2011] [error] [client 10.9.100.251] PHP Notice: 
Undefined variable: set_end in
/var/www/opensips-cp/web/tools/system/siptrace/tracer.php on line 106,
referer:
http://10.9.101.166/cp/tools/system/siptrace/tracer.php?action=search
[Thu Dec 01 23:42:55 2011] [error] [client 10.9.100.251] File does not
exist: /var/www/favicon.ico 

These From opensips.log when I press Search or  Show all :

Dec  1 23:43:53 opensips /sbin/opensips[10425]: DBG:mi_fifo:mi_fifo_server:
done parsing the mi tree
Dec  1 23:43:56 opensips /sbin/opensips[10425]: DBG:mi_fifo:mi_parse_node:
end of input tree



Any advice?



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Re: [OpenSIPS-Users] Modify To:'s field and forward

[spady]

Hi Bogdan, thanks for your advice.
I was wondering what will happen to real RTP stream. I mean: as I sayd, for
Fax server point of view, OpenSIPS acts as Sip Proxy to direct the T38 flow,
after correct fax's handshake. 
OpenSIPS, after analyzed/changed the INVITE, should redirect all stream to
IP PBX. Can OpenSIPS do that? or I have to install rtpproxy module? or
something like that?

Sorry for dummy question but I am really new with OpenSIPS.

Regards

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[OpenSIPS-Users] Modify To:'s field and forward

[spady]

Hi all, I was wondering if OpenSIPS can help me with scenario explained
below;

I have a Fax Server SIP and T38 based. It use a SIP based Pbx as PSTN
Gateway, so requested external fax number are sent from Fax server toward
PBX and then forwarded to Pstn.

Here there's a full SIP trace:
NOTE:
172.16.52.7 = Fax Server
10.9.6.3= PBX 


INVITE sip:00522375568@10.9.6.3:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.52.7:5061
*From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=B0A3C63723A2*
To: sip:00522375568@10.9.6.3:5060
Call-ID: 8f917371-ff97-4953-bc3d-fdba788fe45a@172.16.52.7
CSeq: 101 INVITE
Max-Forwards: 70
Contact: sip:+390522375507@172.16.52.7:5061;user=phone
User-Agent: Alcatel-Lucent OmniTouch Fax Server Application/6.5.6.28
P-Alcatel-CSBU: charging=sip:2542@ucalcatel.sedoc.locale
Content-Type: application/sdp
Content-Length: 235

v=0
o=XMedius-Fax-Gateway 79844629 629 IN IP4 172.16.52.7
s=SIP Fax Call
c=IN IP4 172.16.52.7
t=0 0
m=audio 62186 RTP/AVP 18 8 4
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=ptime:20
SIP/2.0 100 Trying
To: sip:00522375568@10.9.6.3:5060
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=B0A3C63723A2
Call-ID: 8f917371-ff97-4953-bc3d-fdba788fe45a@172.16.52.7
CSeq: 101 INVITE
Via: SIP/2.0/UDP 172.16.52.7:5061
Content-Length: 0

SIP/2.0 180 Ringing
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE
Contact: sip:10.9.6.3
User-Agent: OmniPCX Enterprise R9.0 h1.301.50
P-Alcatel-CSBU: categparty=external
Content-Type: application/sdp
To: sip:00522375568@10.9.6.3:5060;tag=4d23b434bddc6d2bbdcdeb5b281baf8e
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=B0A3C63723A2
Call-ID: 8f917371-ff97-4953-bc3d-fdba788fe45a@172.16.52.7
CSeq: 101 INVITE
Via: SIP/2.0/UDP 172.16.52.7:5061
Content-Length: 229

v=0
o=OXE 1322519520 1322519520 IN IP4 10.9.6.3
s=abs
c=IN IP4 10.9.6.8
t=0 0
m=audio 32600 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:30
a=maxptime:40
a=rtpmap:96 telephone-event/8000
a=sendrecv
SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE
Contact: sip:10.9.6.3
Supported: replaces,timer,100rel
User-Agent: OmniPCX Enterprise R9.0 h1.301.50
Session-Expires: 1800;refresher=uas
P-Alcatel-CSBU: categparty=external
P-Asserted-Identity: Lw RE sip:05223755@10.9.6.3;user=phone
Content-Type: application/sdp
To: sip:00522375568@10.9.6.3:5060;tag=4d23b434bddc6d2bbdcdeb5b281baf8e
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=B0A3C63723A2
Call-ID: 8f917371-ff97-4953-bc3d-fdba788fe45a@172.16.52.7
CSeq: 101 INVITE
Via: SIP/2.0/UDP 172.16.52.7:5061
Content-Length: 229

v=0
o=OXE 1322519520 1322519521 IN IP4 10.9.6.3
s=abs
c=IN IP4 10.9.6.8
t=0 0
m=audio 32600 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:30
a=maxptime:40
a=rtpmap:96 telephone-event/8000
a=sendrecv
ACK sip:10.9.6.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.52.7:5061
From: Fax Alcatel sip:+390522375507@172.16.52.7:5061;tag=B0A3C63723A2
To: sip:00522375568@10.9.6.3:5060;tag=4d23b434bddc6d2bbdcdeb5b281baf8e
Call-ID: 8f917371-ff97-4953-bc3d-fdba788fe45a@172.16.52.7
Max-Forwards: 70
CSeq: 101 ACK
Contact: sip:+390522375507@172.16.52.7:5061;user=phone
Content-Length: 0

INVITE sip:+390522375507@172.16.52.7:5061;user=phone SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE
Contact: sip:10.9.6.3
Supported: replaces,timer,100rel
User-Agent: OmniPCX Enterprise R9.0 h1.301.50
Session-Expires: 1800;refresher=uac
Min-SE: 900
Content-Type: application/sdp
To: sip:+390522375507@172.16.52.7:5061;tag=B0A3C63723A2
From: sip:00522375568@10.9.6.3:5060;tag=4d23b434bddc6d2bbdcdeb5b281baf8e
Call-ID: 8f917371-ff97-4953-bc3d-fdba788fe45a@172.16.52.7
CSeq: 2033886164 INVITE
Via: SIP/2.0/UDP 10.9.6.3;branch=z9hG4bK3dec7cf801610f04d0d388ebc2906678
Max-Forwards: 70
Content-Length: 268

v=0
o=OXE 1322519520 1322519522 IN IP4 10.9.6.3
s=abs
c=IN IP4 10.9.6.8
t=0 0
m=image 32603 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:256
a=T38FaxMaxDatagram:512
a=T38FaxUdpEC:t38UDPRedundancy
INVITE sip:+390522375507@172.16.52.7:5061;user=phone SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE
Contact: sip:10.9.6.3
Supported: replaces,timer,100rel
User-Agent: OmniPCX Enterprise R9.0 h1.301.50
Session-Expires: 1800;refresher=uac
Min-SE: 900
Content-Type: application/sdp
To: sip:+390522375507@172.16.52.7:5061;tag=B0A3C63723A2
From: sip:00522375568@10.9.6.3:5060;tag=4d23b434bddc6d2bbdcdeb5b281baf8e
Call-ID: 8f917371-ff97-4953-bc3d-fdba788fe45a@172.16.52.7
CSeq: 2033886164 INVITE
Via: SIP/2.0/UDP 10.9.6.3;branch=z9hG4bK3dec7cf801610f04d0d388ebc2906678
Max-Forwards: 70
Content-Length: 268

v=0
o=OXE 1322519520 1322519522 IN IP4 10.9.6.3
s=abs
c=IN IP4 10.9.6.8
t=0 0
m=image 32603 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF

Re: [OpenSIPS-Users] Modify To:'s field and forward

[spady]

Hi Osiris, thanks for reply.
I need to modify TO field because this Fax Server has to be used by several
companies. Each company should use own lines. This Fax server cannot do that
for example by setting a profile and assign to that profile a specific
outbound lines.. So, what i should do, is pass to PBX a prefix ( es: 87019 )
that is specific for a outbound line. ( 0 is a common prefix for all... so
I cannot pass it to PBX. In this case will be used always same lines. Hope
it's more clear now ).

Btw  what [hidden mail] stand for? 
So done this, I will have a new INVITE with TO modified, right? 
Mmmm I realized right now that the real SIP trace will not be like posted.
What I posted is taken without OPENSIPS. In my scenario, into TO filed there
will be,for istance, 
To: sip:00522375568@10.9.6.3:5060 but  10.9.6.3 is OpenSIPS server, and
not PBX.
Can I also change it with PBX's IP because at the end the fax call has
to be processed by PBX ( T38 stream ).

Best regards

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